Re: [asterisk-users] [SPAM] - Re: queue moh - Email found in subject
Hi Ioan, I have done that [Set(CHANNEL(musicclass)=…] but it still doesn’t work when a ‘queue’ call is put on hold. So, I can get it to play the correct moh when I use the ‘r’ option – but still get silence when I don’t include the ‘r’. So I’ve sort of fixed my second point (with the ‘r’) but the first point (without the ‘r’) is still not working. Thanks for your feedback ☺ Cheers Andy From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ioan Indreias Sent: 10 July 2013 20:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [SPAM] - Re: [asterisk-users] queue moh - Email found in subject Hello Andy, Have you tried using SetMusicOnHold command before Queue command? BR, Ioan On Wed, Jul 10, 2013 at 7:55 PM, Andrew Thomas a...@datavox.co.uk wrote: Hi All, Sorry if this has been covered already, but I don't tend to follow this list as close as I should these days. Problem is that if a call comes in to a queue without option 'r' specified - moh plays as expected. Now, when that call is answered, all is fine. Trouble comes when that person then puts the caller on-hold. No moh is heard by the caller (in fact, they get silence). If I use 'r' - then ringing is heard - but the queue's musiconhold/musicclass is ignored completely. When the caller is put on hold, they do hear moh but the default moh context is used - not the moh of the queue. What I need is for the queue's moh to be used when the caller is put on hold (and without using the 'r' feature). Is this possible? * 1.8.16.0 (tried on various flavours of 1.8). Queue static and realtime (same outcome). Cheers Andy -- If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue moh
Hi All, Sorry if this has been covered already, but I don't tend to follow this list as close as I should these days. Problem is that if a call comes in to a queue without option 'r' specified - moh plays as expected. Now, when that call is answered, all is fine. Trouble comes when that person then puts the caller on-hold. No moh is heard by the caller (in fact, they get silence). If I use 'r' - then ringing is heard - but the queue's musiconhold/musicclass is ignored completely. When the caller is put on hold, they do hear moh but the default moh context is used - not the moh of the queue. What I need is for the queue's moh to be used when the caller is put on hold (and without using the 'r' feature). Is this possible? * 1.8.16.0 (tried on various flavours of 1.8). Queue static and realtime (same outcome). Cheers Andy -- If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10.0.0-rc1: dahdi doesn't see card
The Debian command I use is: apt-get install linux-headers-`uname -r` That will get the bits you need and place them in /usr/src/. If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse)
This is a brilliant idea. How do I contribute my attackers to this
list?
Cheers
Andy
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Huddleston
Sent: 22 September 2011 16:11
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP
Abuse)
Sounds like a great idea.. Hopefully the page/account never gets hacked
and bad IP's published.. I could see a great hack of
127.0.0.1
192.168.0.0/16
10.0.0.0/8
getting up there somehow and next thing you know - BAM!
But I haven't RTFM - I'm guessing there is probably a white list that
supersedes the naughty list.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, September 22, 2011 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP
Abuse)
very cool!
On Thu, Sep 22, 2011 at 10:37 AM, J. Oquendo aster...@tormenting.net
wrote:
Apologies for cross posting but some of us aren't on the other list
(vice/versa) and thought both groups would benefit.
For those familiar with the VoIP Abuse Project, no need to explain the
gist of this. I got tired of parsing through the alerts (lists) I
receive via email daily. They're long and sometimes I don't have the
time to post them all. So for now, posting VoIP Abuse addresses straight
to Twitter.
So, anyone trying to compromise a pbx, is now autoposted on an hourly
basis to Twitter. Still working on pulling, have about 4 machines linked
up now, will mop em up during the week.
http://twitter.com/#!/voipabuse
Now, you can concoct a quick script off of it, e.g.:
links -dump http://twitter.com/voipabuse;|awk '/attacker/{print
iptables -A INPUT -s $2 -j DROP| sort -u}'
Will get a quickie soon from my Acme's, nCites, etc. when I have time.
For those NOT familiar with it, please Google it as I don't feel like
typing anymore ;) (sorry)
--
=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT, RWSP, GREM
It takes 20 years to build a reputation and five minutes to
ruin it. If you think about that, you'll do things
differently. - Warren Buffett
42B0 5A53 6505 6638 44BB 3943 2BF7 D83F 210A 95AF
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x2BF7D83F210A95AF
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Re: [asterisk-users] [Fwd: FW: realtime mysql - p4]
Sorry - I meant extconfig.conf - not cdr_mysql.conf (my mistake). I use (and done for a long time) mySQL for realtime storage - and it's never let me down (touch wood). Cheers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: 22 May 2011 22:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [Fwd: FW: realtime mysql - p4] On Fri, 2011-05-20 at 10:05 +0100, Andrew Thomas wrote: Post your cdr_mysql.conf and res_mysql.conf and we'll take it from there. Don't forget to remove any 'private' info first (like passwords). Cheers Tnx for the offer, Wil get the files when got back at the office. I presume that cdr_mysql.conf is only relevant for storing call-data-records? Perhaps that is something for later on. For now, i have to show a working *, with all sip-details in a mysql-DB. Other people pointed out that other means (postgres, ldap) might work better, but that's not an option for me. Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Fwd: FW: realtime mysql - p4]
Post your cdr_mysql.conf and res_mysql.conf and we'll take it from there. Don't forget to remove any 'private' info first (like passwords). Cheers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: 19 May 2011 23:14 To: asterisk-users@lists.digium.com Cc: j.witvl...@mindef.nl Subject: [asterisk-users] [Fwd: FW: realtime mysql - p4] Ok, i tried the suggestion: Instead of: sippuser = resource, database_name, table_name sippeer = resource, database_name, table_name I put in: sippuser = resource, context, table_name sippeer = resource, context, table_name Unfortunately, with the same results. btw i tried both general as default Besids the commands i tried below, isn't there any other way to see what's going on? Perhaps it is totally unrelated, but if i perform a mysql-login on the prompt, i first have to select the database manualy, ie it isn't selected by default for the created mysqluser [in this case: voipadmin] Other wild idea, is there a minimum number of fields that haved to be filled? And why is asterisk complaining about not being able to find the databse, when trying to fill it from the asterisk-CLI? My database _is_ named asterisk.. kc3054*CLI realtime update sipusers set SET port = 4343 WHERE name = 0277611 Failed to update. Check the debug log for possible SQL related entries. Command 'realtime update sipusers set SET port = 4343 WHERE name = 0277611' failed. [May 18 18:47:16] WARNING[16718]: res_config_mysql.c:559 update_mysql: MySQL RealTime: Invalid database specified: 'asterisk' (check res_mysql.conf) I mean, is that silly or what? # grep mysql extconfig.conf |grep sip ;sipusers = mysql,asterisk,sip_devices ;sippeers = mysql,asterisk,sip_devices ;sipusers = mysql,general,sip_devices ;sippeers = mysql,general,sip_devices sipusers = mysql,default,sip_devices sippeers = mysql,default,sip_devices kc3054*CLI module show like mysql Module Description Use Count cdr_mysql.so MySQL CDR Backend 0 res_config_mysql.soMySQL RealTime Configuration Driver 0 app_mysql.so Simple Mysql Interface 0 3 modules loaded kc3054*CLI kc3054*CLI sip show users Username Secret Accountcode Def.Context ACL ForcerPort j.witvliet geheimdefault No Yes 027761125b06d3a0b5ef73 default No Yes kc3054*CLI kc3054*CLI sip show peers Name/username HostDyn Forcerport ACL Port Status Realtime 0277611(Unspecified)D N 0Unmonitored j.witvliet (Unspecified)D N 0Unmonitored 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] kc3054*CLI kc3054*CLI kc3054*CLI kc3054*CLI realtime mysql cache kc3054*CLI realtime mysql status general connected to asterisk@127.0.0.1, port 3306 with username voipadmin for 18 seconds. kc3054*CLI -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options
Re: [asterisk-users] Light indicator managed by Asterisk
I would think that that is down to either your indications.conf (could
be wrong) or the handset itself.
I know most Yealink and GrandStream handsets let you change tones in
their individual config. Not too sure about others.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 16 May 2011 17:18
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Light indicator managed by Asterisk
Hello,
this light indicator thing is working just great by following the same
guide as BLF (with hints).
There is just 1 thing bothering me : it is a call that is being made to
an extension, which Asterisk immediately hangs up. This makes the
IP-phone go beep beep beep beep, a normal ringtone when the other end
(Asterisk) has terminated the call.
But is there a way to give a signal to the phone that the line has not
been disconnected so it does not make this annoying beep beep beep
beep sound ? Perhaps a stupid question...
This is my dialplan :
exten = ,1,NoOp(devstate)
exten = ,n,Answer()
exten =
,n,GoToIf($[${DEVICE_STATE(Custom:light)}=BUSY]?unbusy:busy)
exten = ,n(busy),Set(DEVICE_STATE(Custom:light)=BUSY)
exten = ,n,Hangup()
exten = ,n(unbusy),Set(DEVICE_STATE(Custom:light)=NOT_INUSE)
exten = ,n,Hangup()
After the Hangup(), the IP-phone goes beep beep beep beep indicating
the call has ended. I should be glad with this ringtone signal, but not
in this case.
Kind regards,
Jonas.
On 05/12/2011 07:34 PM, Eric Wieling wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jonas Kellens
Sent: Thursday, May 12, 2011 1:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Light indicator managed by Asterisk
On 05/12/2011 07:12 PM, Carlos Chavez wrote:
On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote:
Hello,
is there some way to make Asterisk light up a certain light on an
IP-phone ?
Like MWI, the message waiting indicator can light up if there is
voicemail.
Could this light, or even other lights (like BLF-buttons)
be used to
give a visual notification to the user ?
For example : if a certain value is set in the Mysql-DB
and Asterisk
reads out this value, can Asterisk react upon it inside
the dialplan
to make a light lit up ?
2nd example : if a certain extension is called, can we
perform inside
the dialplan an action that makes a light lit up on a Snom
or Yealink
IP-phone ?
I don't know if all this is at all possible, but it doesn't harm
asking I guess...
If BLF works, then maybe more things are possible in the same way.
Just thinking outside the box here.
BLF lights can be manipulated with Hints and the
DEVSTATE function to
set custom device states. This way you can have a BLF
light react to
any event you want.
This means that extensions/hints need to be defined to be able to
control a BLF-light that monitors this extension ?
I agree that this gives some control over a light/button on
an IP-phone.
The MWI can be manipulated in several ways. Last week
someone asked
this question and got several answers.
You don't perhaps have a link to the discussion ? I don't
really follow
this list constantly so I've certainly missed out on this subject.
pbx*CLI core show application minivmmwi
-= Info about application 'MinivmMWI' =-
[Synopsis]
Send Message Waiting Notification to subscriber(s) of mailbox.
[Description]
This application is part of the Mini-Voicemail system, configured in
min
ivm.conf.
MinivmMWI is used to send message waiting indication to any devices
whose
channels have subscribed to the mailbox passed in the first parameter.
[Syntax]
MinivmMWI(username@domain,urgent,new,old)
[Arguments]
username
Voicemail username
domain
Voicemail domain
urgent
Number of urgent messages in mailbox.
new
Number of new messages in mailbox.
old
Number of old messages in mailbox.
[See Also]
Not available
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Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not
And why would you post a reply 5 days after my last post - and 4 days after the threads last one? Do you want to keep this thread going? I suggest letting it die on it's own. _ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: 17 May 2011 02:05 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not Seriously guys. Why would anyone other than the two of you need to read this. It's a personal conversation. We all know who you both are and your achievements etc. The longer the conversation goes on the more off topic it becomes :-) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI Error
This sounds like you have it set for T1 somehow? Have you upgraded anything lately? Other than that, a Trend tester will show the problem(s) to you. BTW - E1's are 32 channel (not 31). It's 30B+2D. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup Sent: 13 May 2011 16:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI Error I can dial 1-24 channels but not after that. There are 8 E1s. Box was working fine and carrying traffic on all E1s before. Just recently i noticed this problem has occurred. On 13 May 2011 16:30, Rafael Visser visser.raf...@gmail.com wrote: I didn't understand very well.. So you cant dial on the first 24 channels? Did you take care on the jumper of the card?. There is something related to E1 (31 channels) or T1 (24 channels). And check the system.conf either. rv 2011/5/13 deeps backup backup.de...@gmail.com I have checked destination numbers are correct as otherwise calls to those numbers are connecting fine. I opened verbose logs and digged into it more. I found out can't dial any channels from DAHDI/24 on first E1. Before that channel calls are going through fine. I tried test calls to second E1 and can't dial on it either. When I check channel or E1 status it is showing fine. Checked chan_dahdi and system conf files and see all channels are configured fine. Could you please help? On 13 May 2011 15:07, deeps backup backup.de...@gmail.com wrote: On 13 May 2011 14:06, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup Sent: Friday, May 13, 2011 9:02 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI Error Hi, Sometimes calls on Asterisk fail to connect to DAHDI channels and giving below error: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) There are 8 E1 connected on server and only 15-20 simultaneous calls. All channels and E1 are showing in service without any alarms. Could anyone please let me know why this is happening? The message is likely coming from the telco or from the destination number. It is a common issue. I usually put something in my dialplan to retry all calls that receive an unexpected hangup cause to work around the telco seemingly randomly sending back odd hangup causes. You should not retry ALL calls, only ones with unexpected hangup causes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have checked destination numbers are correct as otherwise calls to those numbers are connecting fine. I opened verbose logs and digged into it more. I found out can't dial any channels from DAHDI/24 on first E1. Before that channel calls are going through fine. I tried test calls to second E1 and can't dial on it either. When I check channel or E1 status it is showing fine. Checked chan_dahdi and system conf files and see all channels are configured fine. Could you please help? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has
Re: [asterisk-users] Backport of DEVICE_STATE to 1.4
https://issues.asterisk.org/view.php?id=15818 That's where I get it from. If it contains errors, then why not report it there? Cheers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 13 May 2011 15:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Backport of DEVICE_STATE to 1.4 Hi, Here http://www.voip-info.org/wiki/view/Asterisk+func+device_State you can find a link to download a backported for Asterisk 1.4 version of DEVICE_STATE function. (Elsewhere, you can find reference to another backported function DEVSTATE which seems to behave the same as DEVICE_STATE). As I would like to prepare as much as possible, my dialplan to 1.6 and beyond, I would prefer to use DEVICE_STATE if possible. Anyway, a quick inside this fucn_devstate.c file shows that some (all ?) Log or Error messages are still refering to DEVSTATE. My question is which is the best source to get DEVICE_STATE function for Asterisk 1.4 ? Regards If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Backport of DEVICE_STATE to 1.4
Ah! Forgot about that.
Looks like your on your own Olivier.
Sorry
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif
Madsen
Sent: 16 May 2011 13:12
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Backport of DEVICE_STATE to 1.4
On 11-05-16 07:29 AM, Olivier wrote:
As this bug is considered fixed, I think you can't add any comment
anymore. Unfortunately, you can still see lines mentionning DEVSTATE
function like :
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, DEVSTATE function called with no
custom device
name!\n);
return -1;
}
I opened issue 19300 for that.
Sorry, but backported code is not supported on the issue tracker. You'll
need to use a version of Asterisk that natively supports the
DEVICE_STATE() function and which has maintenance support status (i.e.
Asterisk 1.8).
Thanks,
Leif.
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Every effort has been made to ensure that this e-mail or any attachments
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Re: [asterisk-users] Light indicator managed by Asterisk
Probably using XML - which is phone dependant. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: 12 May 2011 21:18 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Light indicator managed by Asterisk On 05/12/2011 07:12 PM, Carlos Chavez wrote: On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote: Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside the dialplan to make a light lit up ? 2nd example : if a certain extension is called, can we perform inside the dialplan an action that makes a light lit up on a Snom or Yealink IP-phone ? I don't know if all this is at all possible, but it doesn't harm asking I guess... If BLF works, then maybe more things are possible in the same way. Just thinking outside the box here. BLF lights can be manipulated with Hints and the DEVSTATE function to set custom device states. This way you can have a BLF light react to any event you want. Hello, I must say that I have succeeded in working with DEVSTATE to get a BLF-light in several colors. Which works great for what I want. Thank you for the feedback. Do you think it is also possible to get info displayed on the screen of the IP-phone ? Any idea how that would work ? Something tells me that this will depend on the brand/type of IP-phone. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Light indicator managed by Asterisk
Cor-wrong (sort of). There is a backport of DevState/Device_State for 1.4 https://issues.asterisk.org/view.php?id=15818 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: 12 May 2011 20:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Light indicator managed by Asterisk Correct. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, May 12, 2011 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Light indicator managed by Asterisk Eric Wieling wrote: pbx*CLI core show application minivmmwi Core show application minivmmwi core show function DEVICE_STATE Both of these must be a 1.6.x or newer, I have neither under 1.4 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not
[This is my last post in this thread - as I really CBA anymore!] Wow! You really don't see it do you? Fair enough. I thought you were just playing along with my 'ego baiting' game - but it seems I hit the mother load of all ego's here. Apologies to all 'watchers' - but this was intended as a bit of fun - and I thought Steve was just playing along. Seems I was wrong. As a parting gesture though - I'll give you an example from your first post in here (which reads more like a CV/resume than a post! [in fact, they all do]): I was the number 1 poster on this list a couple of years ago I don't really do job searches, I am usually offered a job or project and approached by the client. My last trip was to Iraq, but I have been to Senegal, Sierra Leone, Guinea, Ghana, Liberia to help rebuild the infrastructure for USAID. For the Dept of State, I set up... For DoD/Dos, I cannot really say much except... How many VoIP guys were taking ak47 rounds while I was on top of the Iraqi Government building... A bit further in to the thread: Would you say that I am a productive member of the list and go pretty far out of my way to help people? Most of the time give useful info, like the Outbound Caller ID thread? [fish] And again: I do not email people... - then why did you just e-mail me off list? Your last post: ...because I own thousands of ounces of silver bullion... Everyone a winner [and not one relevant to the thread or the discussion]. Anyway - truth does indeed hurt mate. Grab yourself a Kleenex as I throw you back in to the pond. Goodbye. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: 11 May 2011 17:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not I am not upset in least, well I am but that's because I own thousands of ounces of silver bullion and I am watching in get pummeled again. Good thing I bought the bulk of it when it was only $12 an ounce. http://www.kitco.com/charts/livesilver.html You are an angry person and it is sad. It is also sad that the example I requested earlier is something posted later. The only reason for that is because you had nothing to back up any of your rage. Seek help, please. If you feel like you want to hurt yourself or others, have yourself committed right away. I am serious. If you are voluntary, you can leave when you want. Thanks, Steve Totaro On Wed, May 11, 2011 at 12:13 PM, Andrew Thomas a...@datavox.co.uk wrote: Seems I have upset the God that is Steve Totaro! You want an example? OK - your last post. Has nothing to do with the thread (or our 'discussion') but yet you chose to post it as yet another self pat-on-the-back! I could produce a lot more - but you now bore me. You know it must be so hard being so perfect Steve. I so wish I was you! Have a really nice day :) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: 11 May 2011 16:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not Yeah, I am not sure why dude went on the offensive. Got emotional but could not produce a single example of the name calling and insults he was hurling at me. Here is an email I received a very short time ago. Sender and company's name have been removed. | to Steve show details 9:15 AM (2 hours ago) steve, I haven't been active in the * community for a while but ran across an interesting project that I would like to pursue. [COMPANY] in springfield needs a * admin part time and I could use the steady income plus I would like to get my hands back into *... I wanted to check with you first because this is your neck of the woods... do you have any experience with them? recently, I have been just lurking on the [asterisk-] lists. thanks for supporting the [asterisk-] groups in a big way. On Wed, May 11, 2011 at 11:28 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Wow...somehow this turned into a something so much darker than the original intent*sits back and watches the show* Thanks guys, that little mini bonfire made an otherwise boring day into an entertaining Asterisk-Users version of WWE Raw. Cheers! Sherwood McGowan
Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not
Wow! How self-promoting was that post? As for a simple 'that worked' post - as others have already pointed out before you, it's not for self-gratification - it's to help anyone else who has the same/similar problem. I used the list archives quite a lot in my early days - and having the last post in a thread say 'try this, this or this' and no comeback is a pain. A simple 'option 2 worked for me' post at then end would make everything a lot simpler (and beat those deadlines you talked about). As for 'off-list' mailing - please do NOT do it without asking/permission as most people get enough e-mails as it is (from paying customers). Thanks all and have a nice day! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: 11 May 2011 05:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not On Tue, May 10, 2011 at 8:30 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: +1 from me too. The other thing is that when you answer to say the problem has been solved this goes into the archives meaning that people can use Google to answer their own questions rather than having to even ask the list. There have been times when I've searched for a solution to a problem, found like 10 answers, and nobody has said whether they work or not so you have to try all of them. -- Cheers, Matt Riddell Believe me mate, I feel you, on that note. Not only because of my time when I was asking more questions than I was answering, but also from the standpoint of wishing the answers were a little more prevalent for the searching party to find so that I didn't see s many repeats on the list ;-) Cheers guys! -- Sherwood McGowan Telecommunications and VOIP Consultant -1 Since I was the number 1 poster on this list a couple of years ago, I think I can speak with some authority. I just assume that if that person does not ask any more questions, that they have either solved the problem on their own, or I helped them by giving the answer or steering them to it. I don't need a public or private Thank You When I was posting all the time, I figured the ratio of Thank you emails to silence to be about 20 to 1, maybe as high as 50 to 1. People are busy, under a deadline or whatever, I offer help and do not expect anything in return, not even a thank you. Probably because I have and will be one of those people, although my questions are usually a little over the top for the list or can be pointed to something in bugtracker, I have asked many questions when I was stuck and under an all nighter deadline. I would like to thank anyone out there that has helped me over the many, many years dealing with Asterisk and VoIP. It is a blanket thank you for all times I simply moved onto then next hurdle to get my deliverables out on time and working properly and neglected to post a thank you. Before there was any documentation, voip-info amd this list was my savior. The volume of traffic has fallen to almost nothing over the last year or two. I wonder if Digium could post totals as it did when I was shocked to find my name as the #1 poster. It would be cool to see who is the #1 poster now, but I am more interested in what I perceive to be a huge fall off of posting. It could be my email server, since I was getting notices from the list about excessive email bounces and removing me if I did not click a link. That seems to have stopped, and I don't think it was on my side. Back to getting credit or a thank you. What I have received by answering questions or helping to troubleshoot is worth way more than a thank you. I get some name recognition, paid work, large call centers, Sr Positions in high profile jobs. Enough to make a nice living, whether I am independent or in a salaried position. Asterisk has literally taken me all over the world. My last trip was to Iraq, but I have been to Senegal, Sierra Leone, Guinea, Ghana, Liberia to help rebuild the infrastructure for USAID. I don't really do job searches, I am usually offered a job or project and approached by the client. For the Dept of State, I set up prepaid call centers to answer questions and getting a reservation at the various Embassies about obtaining a visa to come to the US. It is called the US visa Information Service For DoD/Dos, I cannot really say much except I can say is that I am probably one of the few Asterisk people that were issued a Glock and M4, bullet proof vests, armored cars, and a PSD team.. How many VoIP guys were taking ak47 rounds while I was on top of the Iraqi Government building,
Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not
Let's not get in to to pissing contest. I am not new to this list (jfyi - I am also a dCAp). I do know who you are (and couldn't care less anymore). I, also, have paying customers (but don't feel the need to gloat about it in here). I am not pretending to know you - as I don't know you on a personal level (and don't wish to). Sorry that you feel the need to fish for compliments - but you just don't get them like that from me (besides which - you have no bait!). You carry on doing what you do - and I'll carry on doing what I do (without braodcasting it to the community). Oh, and pleae don't trip over your ego on the way out. Have a nice day! If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not
Snore... Now move along please... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: 11 May 2011 14:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not Alex, thanks for the laugh. I have a wireless keyboard and the batteries are dying. I have been lazy and not picked up some AAAs. I have been using spell check to help. At least the wrong word was spelled correctly, lol. Or he is not really reading what I wrote which was along the same lines as everyone else, but nobody has to post their solutions, it would be nice, but it is a moot point. As far as deadlines and taking a little extra time to send solved, yeah, in a war zone, it could get you killed. Thanks, Steve Totaro On Wed, May 11, 2011 at 9:32 AM, Alex Balashov abalas...@evaristesys.com wrote: On 05/11/2011 09:29 AM, Steve Totaro wrote: You must have a reading compression problem. I would love to bzip2 or gzip the reading process. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not
Seems I have upset the God that is Steve Totaro! You want an example? OK - your last post. Has nothing to do with the thread (or our 'discussion') but yet you chose to post it as yet another self pat-on-the-back! I could produce a lot more - but you now bore me. You know it must be so hard being so perfect Steve. I so wish I was you! Have a really nice day :) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: 11 May 2011 16:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not Yeah, I am not sure why dude went on the offensive. Got emotional but could not produce a single example of the name calling and insults he was hurling at me. Here is an email I received a very short time ago. Sender and company's name have been removed. | to Steve show details 9:15 AM (2 hours ago) steve, I haven't been active in the * community for a while but ran across an interesting project that I would like to pursue. [COMPANY] in springfield needs a * admin part time and I could use the steady income plus I would like to get my hands back into *... I wanted to check with you first because this is your neck of the woods... do you have any experience with them? recently, I have been just lurking on the [asterisk-] lists. thanks for supporting the [asterisk-] groups in a big way. On Wed, May 11, 2011 at 11:28 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Wow...somehow this turned into a something so much darker than the original intent*sits back and watches the show* Thanks guys, that little mini bonfire made an otherwise boring day into an entertaining Asterisk-Users version of WWE Raw. Cheers! Sherwood McGowan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OUTBOUND CALLER ID
Try getting rid of '/5001' (line 2 and 4) and try again!
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh
katta
Sent: 10 May 2011 06:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OUTBOUND CALLER ID
sir,
Below configuration i wase made in server . but this is not working.
exten = _90X,1,NoOp(${CALLERID(num)})
exten = _90X/5001,2,Set(CALLERID(name)=44578999)
exten = _90X,3,AGI(agi://127.0.0.1:4577/call_log)
exten = _90X/5001,4,Set(CALLERID(num)=44578999)
exten =
_90X,5,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALL
ERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten = _90X,6,Dial(${TRUNK}/${EXTEN:1},,tTo)
exten = _90X,7,Hangup
On Mon, May 9, 2011 at 8:14 PM, Carlos Rojas crt.ro...@gmail.com
wrote:
Hello
Do you set your callerid in the context outgoing?
[outgoing]
exten = _X.,1,Set(CALLERID(num)=4663000)
exten = _X.,n,Dial(..
On Mon, May 9, 2011 at 4:45 AM, mahesh katta maheshka...@flexydial.com
wrote:
Sir ,
this is not working
On Mon, May 9, 2011 at 1:52 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
On Monday 09 May 2011, mahesh katta wrote:
Hi,
THIS IS IN DUBAI.
I am having PRI line with 100 DID's (00-99) and when we call to any
landline or mobile number then it shows us our board number or pilot
number
(i.e 4663000 means 00)..
In the context through which outgoing calls are placed, you need a step
which
sets the caller ID number. For instance, part of our dialplan maps
external
phone numbers with the local part 707060 to 707072 to internal
extensions 301
to 312 respectively. Our E1 provider also requires us to include the
STD
code, minus the leading zero, for the town we are in -- and will
silently
anonymise the call if we try to send a caller ID that does not belong to
us.
So for outgoing calls, we have something like
[ts-outgoing]
exten = _0., 1, Set(localno=7070$[${CALLERID(num)}-240])
exten = _0., 2, Set(CALLERID(num)=${STD}${localno})
--
AJS
Answers come *after* questions.
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Best Regards,
Mahesh Katta
BUZZWORKS Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
(E) Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
http://www.buzzworks.com/
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Best Regards,
Mahesh Katta
BUZZWORKS Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
(E) Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
http://www.buzzworks.com/
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus
Re: [asterisk-users] OUTBOUND CALLER ID
Why do I get the feeling that this guy wants someone to write it for him
for free?
Especially seeing has how he has never posted what anyone who has tried
to help, have requested.
Maybe Mr. Katta needs to google for 'dcap'?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh
katta
Sent: 10 May 2011 11:49
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OUTBOUND CALLER ID
Sir,
A.J.Stiles
This dialplan is not working . when I called to out of box .
On Tue, May 10, 2011 at 2:00 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
On Tuesday 10 May 2011, mahesh katta wrote:
sir,
Below configuration i wase made in server . but this is not
working.
exten = _90X,1,NoOp(${CALLERID(num)})
exten = _90X/5001,2,Set(CALLERID(name)=44578999)
exten = _90X,3,AGI(agi://127.0.0.1:4577/call_log)
exten = _90X/5001,4,Set(CALLERID(num)=44578999)
exten =
_90X,5,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALL
ERI
DNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0)) exten =
_90X,6,Dial(${TRUNK}/${EXTEN:1},,tTo)
exten = _90X,7,Hangup
OK. Here's what I see going on.
When you dial 90X:
Stage 1: The NoOp() will just write the CALLERID(num) to the
console. (This
initially will be the originating extension number.)
Stage 2: If the originating extension is 5001, the
CALLERID(name) will be set
to 44578999.
Stage 3: Calls an AGI script, presumably to log the call
outside of the CDR
database.
Stage 4: If the originating extension is 5001, the
CALLERID(num) will be set
to 44578999.
Stage 5: Starts a recording.
Stage 6: Passes the dialled number, skipping 1 digit from the
beginning
(i.e. the initial 9 for the outside line), to a Dial() command.
Stage 7: Hangs up.
I'm not at all convinced that this is right, especially as you
are mixing
destination extensions with and without originating extensions.
And, the way
this bit is written, it will only ever set the outgoing caller
ID for
extension 5001.
I think it needs to be more like this. Here, I'm taking an
educated guess
that you want your caller ID to appear on outgoing calls as
445789 followed
by the last 2 digits of the extension number. If this is not
right, you will
have to change it -- or explain exactly how to derive the caller
ID you want
to appear on external phones, from the originating internal
extension, like I
originally asked.
exten = _90X,1,NoOp(${CALLERID(num)})
exten =
_90X,2,Set(outgoing_ident=445789${CALLERID(num):-2})
exten = _90X,3,NoOp(${outgoing_ident})
exten = _90X,4,Set(CALLERID(name)=${outgoing_ident})
exten = _90X,5,AGI(agi://127.0.0.1:4577/call_log)
exten = _90X,6,Set(CALLERID(num)=${outgoing_ident})
exten =
_90X,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALL
ERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|
av(0)V(0))
exten =_90X,8,Dial(${TRUNK}/${EXTEN:1},,tTo)
exten = _90X,9,Hangup
What this will do:
Stage 1: The NoOp() will just write the CALLERID(num) to the
console. (This
initially will be the originating extension number.)
Stage 2: Creates a variable outgoing_ident. This consists of
the string
445789 followed by the last 2 digits of the originating
extension number.
Stage 3: The NoOp() will write the value of ${outgoing_ident}
to the console.
Stage 4: Sets CALLERID(name) to the value we just put into
${outgoing_ident}.
Stage 5: Calls logging AGI script.
Stage 6: Sets CALLERID(num) to the value we just put into
${outgoing_ident}.
This is most likely to be noticed.
Stage 7: Starts recording.
Stage 8: Passes the dialled number, skipping 1 digit from the
beginning, to a
Dial() command.
Stage 9: Hangs up.
Modify stage 2 if necessary to suit exactly how you want your
outgoing ident
to appear. You can take out the NoOp() statements and renumber
appropriately
once it's working as you want it.
Note that if the console seems to show you created the right
ident but it
doesn't appear on phones when you dial out, then either the
format is wrong
or your telco doesn't think you are authorised to use that
ident; this is a
matter you will need to take up with your phone company.
--
Re: [asterisk-users] [OT] Yealink Phones
Under 'Phone' there is a new 'Action URL' section (version 60 of firmware). There are a lot of phone 'events' that can call something else. Eg. if the phone goes off-hook (just off-hook) an 'event' is triggered. This 'event' calls a script (in my case) that changes a BLF on reception to busy (even though it isn't really busy as such). My 'test' script gets fired like this: Off hook http://192.168.1.1/test.php?state=offhookextn=201 On hook http://192.168.1.1/test.php?state=onhookextn=201 There are lots of 'events' you can capture. I haven't gotten around to setting up these events via. auto-provision yet - but that's on my list of to-do's. If you get stuck with the Yealink - feel free to contact me off-list if you think it more suitable. Cheers Andy _ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell Brown Sent: 13 April 2011 19:05 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] [OT] Yealink Phones Quoth Andrew Thomas:- Have you seen the 'Action URL' bit yet? Makes everything almost key-system like ;) I saw it in the DSS key settings but havn't worked out anything useful to do with it yet? What are you using it for (and how?)? -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime SIP peer status
Maybe I should have asked 'why do you want to put the status in to a mySQL database'? BTW - extensions.conf has mySQL functions built in - so no external script is actually needed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: 13 April 2011 10:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime SIP peer status On 04/13/2011 11:20 AM, Ishfaq Malik wrote: On Wed, 2011-04-13 at 11:09 +0200, Jonas Kellens wrote: On 04/13/2011 10:57 AM, Ishfaq Malik wrote: On Wed, 2011-04-13 at 10:15 +0200, Jonas Kellens wrote: Hello, I'm using SIP realtime with MySQL DB. Is it possible to get the status of the SIP peer (free / calling) from this realtime DB ? If not, is there another way to obtain the call state of a SIP peer ? You could use core show channels in the console/via AMI to determine if any extensions are on a call or even making a call. If this information is not available, then I'm thinking of writing an AGI and calling this AGI when a call is being answered. This AGI will then write to the MySQL-DB the state busy for this SIP peer. Off course when the call ends, I need another AGi in the h-exten which writes the state free for this SIP peer. You think this will work ? Or will it put too much load on my system ? Kind regards, Jonas. You could write a shell script to do what you suggested and pop it on a cron. The info wouldn't be 100% realtime that way though but I think the load would be very low. Also, as someone else has suggested, you could use hints but you have to add some of the code for hints directly into the extensions.conf which sort of goes against the point of RealTime unless you use scripts to handle that part as I myself have done. Why should I use a cron ? I can just use an AGI in extensions.conf. That's the closest to realtime I think. How can I write information to a MySQL-DB using hints ? Please explain. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime SIP peer status
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL And yes, I meant Asterisk has mySQL commands built in [that can be accessed via. extensions.conf]. Sorry if I mislead. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: 13 April 2011 10:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime SIP peer status On Wed, 2011-04-13 at 10:28 +0100, Andrew Thomas wrote: BTW - extensions.conf has mySQL functions built in - so no external script is actually needed. Could you point me in the right direction for that? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime SIP peer status
Fair enough. Then if this is really what you want I guess an AGI is the best way to go. As for load - well, that depends on how many concurrent connections you figure on having [and of course the platform it's all on]. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: 13 April 2011 10:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime SIP peer status On 04/13/2011 11:28 AM, Andrew Thomas wrote: Maybe I should have asked 'why do you want to put the status in to a mySQL database'? BTW - extensions.conf has mySQL functions built in - so no external script is actually needed. Well, I read out this information in a website which serves as a comprehensible GUI. I know I can use mysql-functions in the dialplan, but when I need to write something on answering, then I need the AGI-option of the Dial()-command. Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Yealink Phones
Hi Russell,
Have you seen the 'Action URL' bit yet? Makes everything almost
key-system like ;)
BTW - one downfall of the Yealink is that it can't send different DND
commands to different accounts (it sends the one command to all
accounts). Not very useful if providers use different commands for DND
(like they tend to). I know Yealink are working on this though - as I
am one of the 'beta' testers.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell
Brown
Sent: 13 April 2011 10:02
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [OT] Yealink Phones
I've just started deploying these (well the T28P model) after years of
Snom issues and they look pretty good (although the documentation is
execrable; if you thought the Snom stuff was obtuse Yealink have got
them knocked into a cocked hat!).
Anyway, for provisioning I use HTTP with a DHCP entry like:-
#
# Yealink Phones
#
group {
#
# The phone should pickup the
# model config file (y0.cfg for the
# T28P) first and then the MAC.cfg file
#
# Yes tftp-server-name to set the DHCP
option but
# the http:// tells the phone to get
it's files via
# http.
option tftp-server-name
http://192.168.1.13/yealink;;
#
host yealinkT28P {
hardware ethernet 00:15:65:1b:d9:12;
fixed-address 192.168.1.33;
option host-name yealinkT28P;
}
}
As the comments say, the phone's first pick up the model dependant
config file (y0.cfg for the T28P model) and then the MAC.cfg
file.
This is nice as you have one model.cfg file for the site-wide config and
then fine tune specific phones (setup different BLF keys and, obviously,
SIP logins for each device) in the MAC.cfg files.
In the y0.cfg file I have:-
#
# Auto Provision
[ autoprovision ]
path = /config/Setting/autop.cfg
server_address = http://192.168.1.13/yealink
[ autop_mode ]
path = /config/Setting/autop.cfg
# Mode 7 = at Power On and Weekly
mode = 7
# Sunday between 0100 and 0500
schedule_dayofweek=0
schedule_time = 01:00
schedule_time_end = 05:00
#
Re non-web based access. Obviously the config files are on your
DHCP/Apache/Asterisk server so you can edit them however you like.
You can also enable telnet access to the phones with a 'hidden' config
option of:-
#
[ telnet ]
path=/config/Network/Network.cfg
telnet_enable=1
#
but the login/password are the admin defaults so a bit of a security
hole there. Not really found much useful telnetting into the phone but
I've not played around with it much.
One other useful tip: If you play around in the web interface, set the
phone up and then export the config, you end up with a config.bin file
which is just tar of the config files. A quick diff and you can easily
find out what you need to tweak in your Autoprovision config files.
Hope that helps.
PS - anyone else with useful Yealink tips?
--
Regards,
Russell
| Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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Re: [asterisk-users] send voicemail to multiple emails
So why not simply go back to square one and create a 'distribution group' e-mail address - and send to that? You've probably realised by now that if you want * to do something it doesn't already do - you have to write that bit yourself. Good luck. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: 11 April 2011 13:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] send voicemail to multiple emails We are talking about mailcmd not externnotify I am aware of extennotify, problem is, it runs script when someone checks their voicemail, i need a script to run only when a voicemail is left On Mon, Apr 11, 2011 at 6:32 AM, Andrew Thomas a...@datavox.co.uk wrote: Not quite true. I use a PHP script to do my processing (called from voicemail.conf [externnotify = /usr/local/bin/vmnotify.php]). The main three lines are: $vm_context = $argv[1]; $extension = $argv[2]; $number_of_messages = $argv[3]; Self explanatory really. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: 10 April 2011 05:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] send voicemail to multiple emails I've already taken the steps you described...issue i ran into was there is no variables passed to mailcmd only STDIN... as a result i have to extract variables from STDIN... On Fri, Apr 8, 2011 at 5:09 PM, Warren Selby wcse...@selbytech.com wrote: On Fri, Apr 8, 2011 at 1:18 PM, vip killa vipki...@gmail.com wrote: That does not sound easy... besides these email addresses would be taken from a MySQL database. It's actually what you're going to end up doing, whether you do it on the MTA level or your code it into your script that you execute instead of sendmail -f. Currently, there is no way to natively have asterisk send one voicemail to multiple email addresses. What's probably going to work best for you since you seem to like program your own scripts (and I'm not talking an AGI here, I'm talking either pure bash, php, perl, or whichever you prefer), is to change the mailcmd= option inside voicemail.conf and replace it with a script of your own design. I'm not sure off the top of my head which variables are passed to the command, but you could always write a simple script that just outputs all arguments to see and go from there. My guess is you're going to at the least get the preconfigured email address and the contents of your emailsubject and emailbody options (both of which have the option of passing multiple useful variables). -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman
Re: [asterisk-users] send voicemail to multiple emails
Not quite true. I use a PHP script to do my processing (called from voicemail.conf [externnotify = /usr/local/bin/vmnotify.php]). The main three lines are: $vm_context = $argv[1]; $extension = $argv[2]; $number_of_messages = $argv[3]; Self explanatory really. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: 10 April 2011 05:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] send voicemail to multiple emails I've already taken the steps you described...issue i ran into was there is no variables passed to mailcmd only STDIN... as a result i have to extract variables from STDIN... On Fri, Apr 8, 2011 at 5:09 PM, Warren Selby wcse...@selbytech.com wrote: On Fri, Apr 8, 2011 at 1:18 PM, vip killa vipki...@gmail.com wrote: That does not sound easy... besides these email addresses would be taken from a MySQL database. It's actually what you're going to end up doing, whether you do it on the MTA level or your code it into your script that you execute instead of sendmail -f. Currently, there is no way to natively have asterisk send one voicemail to multiple email addresses. What's probably going to work best for you since you seem to like program your own scripts (and I'm not talking an AGI here, I'm talking either pure bash, php, perl, or whichever you prefer), is to change the mailcmd= option inside voicemail.conf and replace it with a script of your own design. I'm not sure off the top of my head which variables are passed to the command, but you could always write a simple script that just outputs all arguments to see and go from there. My guess is you're going to at the least get the preconfigured email address and the contents of your emailsubject and emailbody options (both of which have the option of passing multiple useful variables). -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BRI detection
NT = Network Termination/Topology (or something like that) - used when you want to be the network end. TE = Terminating Equipemt - used when you want to be the consumer end (a PBX or ISDN handset usually). You probably want to be the TE - as you are running Asterisk PBX ;) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta Sent: 01 April 2011 13:37 To: asterisk-users@lists.digium.com Subject: [asterisk-users] BRI detection Hi, I need to configure BRI 4span card in dubai in vicidialnow for dialer perpose. in that i have small confusion which is NT an TE mode . that was i am setting perfectly but dubai telco what they are use for this i dont know which parameters are use for that . please help me. -- Best Regards, Mahesh Katta BUZZWORKS Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 303, Gagangiri Apts, Parleshwar Road, Ville Parle East, Mumbai - 400057. GSM +91.97029.70779 | Phone +91.22.2663.1811 | Fax +91.22.2663.1811 Web http://www.buzzworks.com If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
Just to respond to the IP range approach. My ISP recently changed my external IP and now it appears that I am in New York (when I am actually static in Manchester, England). I've also been in Birmingham, Motherwell and Nottingham [UK] aswell! So, although banning certain ranges may be a good idea for you - it's not a good idea for everyone (we have 'road warriors' that do, indeed, travel to the Far East and Middle East). I suppose the only 'real' way to invoke security (on any system) is to have very strong passwords - maybe 1234 is not the way to go :p -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: 30 March 2011 10:08 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk and fail2ban On Wed, 30 Mar 2011 01:45:20 +0300, Ioan Indreias indre...@gmail.com wrote: Just to provide an alternative to sshguard: you could use BFD[1] Thanks Ioan. I'll give it a shot. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem routing call to fax machine on DAHDI FXSport
[18884732963@from-fax-machine:... - your call is hitting the
from-fax-machine context - yet your 'fax' exten is in the from-pstn-4
context. See the [2011-03-17 13:40:29.6] NOTICE[8825] chan_dahdi.c:
Fax detected, but no fax extension line.
When Asterisk detects an incoming fax tone - it tries to automagically
route the call to the 'fax' extension in the SAME context as the
incoming call.
Turning the fax detect off will cure this - but you will lose auto fax
detection.
I suggest adding:
[from--fax-machine]
...
exten = fax,1,Goto(from-pstn-4,fax,1)
I actually have a completely separate context for incoming faxes - and
just send any detections straight to it (using the above method).
HTH
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank
Tarczynski
Sent: 18 March 2011 03:03
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem routing call to fax machine on DAHDI
FXSport
I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS
modules. I'm trying to set-up things to route analog fax calls from a
FXO port to an analog fax machine on a FXS port on the same card.
Outgoing faxes work just fine. But incoming faces are routed to the
right DAHDI extension, but the call dropped right as the fax machine
rings for the first time. The fax machine works fine when connected
directly to the analog telephone line and I see the same behavior if I
route the fax call to anyother DAHDI or SIP extension.
Can anyone help?
I see this in the asterisk log:
(Send fax out to HP's fax check line)
[2011-03-17 13:40:17.4] VERBOSE[8825] chan_dahdi.c: -- Starting
simple switch on 'DAHDI/1-1'
[2011-03-17 13:40:24.0] VERBOSE[8825] pbx.c: -- Executing
[18884732963@from-fax-machine:1] Set(DAHDI/1-1,
CALLERID(num)=19195718465) in new stack
[2011-03-17 13:40:24.0] VERBOSE[8825] pbx.c: -- Executing
[18884732963@from-fax-machine:2] Dial(DAHDI/1-1,
DAHDI/4/18884732963) in new stack
[2011-03-17 13:40:24.0] VERBOSE[8825] app_dial.c: -- Called
4/18884732963
[2011-03-17 13:40:26.2] VERBOSE[8825] app_dial.c: -- DAHDI/4-1
answered DAHDI/1-1
[2011-03-17 13:40:29.6] NOTICE[8825] chan_dahdi.c: Fax detected, but no
fax extension
[2011-03-17 13:41:13.4] VERBOSE[8825] chan_dahdi.c: -- Hungup
'DAHDI/4-1'
[2011-03-17 13:41:13.4] VERBOSE[8825] pbx.c: == Spawn extension
(from-fax-machine, 18884732963, 2) exited non-zero on 'DAHDI/1-1'
[2011-03-17 13:41:13.4] VERBOSE[8825] chan_dahdi.c: -- Hungup
'DAHDI/1-1'
(Incoming fax attempt)
[2011-03-17 13:43:18.3] VERBOSE[8834] chan_dahdi.c: -- Starting
simple switch on 'DAHDI/4-1'
[2011-03-17 13:43:19.3] VERBOSE[8834] pbx.c: -- Executing
[s@from-pstn-4:1] Wait(DAHDI/4-1, 1) in new stack
[2011-03-17 13:43:20.4] VERBOSE[8834] pbx.c: -- Executing
[s@from-pstn-4:2] Verbose(DAHDI/4-1, CALLERID is 8884732963) in new
stack
[2011-03-17 13:43:20.4] VERBOSE[8834] app_verbose.c: CALLERID is
8884732963
[2011-03-17 13:43:20.4] VERBOSE[8834] pbx.c: -- Executing
[s@from-pstn-4:3] Verbose(DAHDI/4-1, Time is 20110317-134320) in new
stack
[2011-03-17 13:43:20.4] VERBOSE[8834] app_verbose.c: Time is
20110317-134320
[2011-03-17 13:43:20.4] VERBOSE[8834] pbx.c: -- Executing
[s@from-pstn-4:4] Answer(DAHDI/4-1, ) in new stack
[2011-03-17 13:43:20.4] VERBOSE[8834] pbx.c: -- Executing
[s@from-pstn-4:5] Ringing(DAHDI/4-1, ) in new stack
[2011-03-17 13:43:20.4] VERBOSE[8834] pbx.c: -- Executing
[s@from-pstn-4:6] Wait(DAHDI/4-1, 6) in new stack
[2011-03-17 13:43:21.4] VERBOSE[8834] chan_dahdi.c: -- Redirecting
DAHDI/4-1 to fax extension
[2011-03-17 13:43:21.4] VERBOSE[8834] pbx.c: == Spawn extension
(from-pstn-4, fax, 1) exited non-zero on 'DAHDI/4-1'
[2011-03-17 13:43:21.4] VERBOSE[8834] pbx.c: -- Executing
[fax@from-pstn-4:1] NoOp(DAHDI/4-1, Fax Detected) in new stack
[2011-03-17 13:43:21.4] VERBOSE[8834] pbx.c: -- Executing
[fax@from-pstn-4:2] Dial(DAHDI/4-1, DAHDI/1,40,tr) in new stack
[2011-03-17 13:43:21.4] VERBOSE[8834] app_dial.c: -- Called 1
[2011-03-17 13:43:21.4] VERBOSE[8834] app_dial.c: -- DAHDI/1-1 is
ringing
[2011-03-17 13:43:23.4] VERBOSE[8834] app_dial.c: -- DAHDI/1-1 is
ringing
(Call is routed to fax machine, but then dropped before it can answer)
[2011-03-17 13:43:24.8] VERBOSE[8834] chan_dahdi.c: -- Hungup
'DAHDI/1-1'
[2011-03-17 13:43:24.8] VERBOSE[8834] pbx.c: == Spawn extension
(from-pstn-4, fax, 2) exited non-zero on 'DAHDI/4-1'
[2011-03-17 13:43:24.8] VERBOSE[8834] chan_dahdi.c: -- Hungup
'DAHDI/4-1'
My dialplan looks like this:
[from-pstn-4]
exten = fax,1,NoOp(Fax Detected)
exten = fax,2,Dial(DAHDI/1,,rtT)
exten = fax,3,Congestion()
exten = fax,104,Busy()
exten = s,1,Wait(1)
exten = s,n,Verbose(CALLERID is ${CALLERID(num)})
exten = s,n,Verbose(Time is ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = s,n,Answer
exten = s,n,Ringing
exten = s,n,Wait(6)
exten =
Re: [asterisk-users] Passing an argument to a macro within an Originatecommand
The last Originate() option is ignored if using 'app'. It is only there
for 'exten'.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Originate tells all :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce
Hopkins
Sent: 15 March 2011 21:36
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Passing an argument to a macro within an
Originatecommand
Hi,
With Asterisk 1.8.3, I can't figure out how to pass an argument to a
macro which is used within an originate command.
Here is my sample dialplan to illustrate:
exten = 123,1,Answer()
exten = 123,n,Originate(SIP/20,app,Macro,foo,bar)
exten = 123,n,NoOp(This is the NoOp after the originate command)
exten = 123,n,Wait(30)
exten = 123,n,Hangup()
[macro-foo]
exten = s,1,Answer()
exten = s,2,NoOp(arg1 is ${ARG1} and arg2 is ${ARG2})
exten = s,3,Playback(tt-monkeys)
I was hoping the ${ARG1} within the macro would be 'bar', but the
argument does not seem to be passed on to the macro so far as I can
tell.
Here is the CLI output:
pbx*CLI
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [123@from-internal:1] Answer(SIP/21-000c, ) in
new stack
-- Executing [123@from-internal:2] Originate(SIP/21-000c,
SIP/20,app,Macro,foo,bar) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Launching Macro(foo) on SIP/20-000d
-- Executing [s@macro-foo:1] Answer(SIP/20-000d, ) in new
stack
-- Executing [s@macro-foo:2] NoOp(SIP/20-000d, arg1 is and
arg2 is ) in new stack
-- Executing [s@macro-foo:3] Playback(SIP/20-000d,
tt-monkeys) in new stack
-- SIP/20-000d Playing 'tt-monkeys.gsm' (language 'en')
-- Executing [123@from-internal:3] NoOp(SIP/21-000c, This is
the NoOp after the originate command) in new stack
-- Executing [123@from-internal:4] Wait(SIP/21-000c, 30) in
new stack
-- Executing [123@from-internal:5] Hangup(SIP/21-000c, ) in
new stack
== Spawn extension (from-internal, 123, 5) exited non-zero on
'SIP/21-000c'
-- Executing [h@from-internal:1] Macro(SIP/21-000c,
hangupcall) in new stack
-- Executing [s@macro-hangupcall:1] GotoIf(SIP/21-000c,
1?skiprg) in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf(SIP/21-000c,
1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf(SIP/21-000c,
1?theend) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup(SIP/21-000c, ) in
new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'SIP/21-000c' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/21-000c'
Could anyone tell me what I am doing wrong please?
Many thanks in advance for any assistance anyone is able to offer.
Best regards
Bruce Hopkins
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 paging with ploycom
[default]
exten = 777,1,Answer()
exten = 777,n,Record(/var/lib/asterisk/sounds/page:gsm)
exten = 777,n,Originate(Local/pb@dv-ip,exten,page-it,s,1)
exten = 777,n,Hangup()
exten = pb,1,Answer()
exten = pb,n,Playback(page)
[page-it]
exten = s,1,Set(page1=SIP/801SIP/802SIP/803) ; etc etc
exten = s,n,SIPAddHeader(Call-Info: \;answer-after=0)
exten = s,n,SIPAddHeader(Answer-Mode: Auto)
exten = s,n,SIPAddHeader(P-Auto-answer: normal)
exten = s,n,SIPAddHeader(Alert-Info: info=alert-autoanswer)
exten = s,n,Page(${page1})
This works for me with a Yealink T28, a Linksys SPA-941, an Aastre 6755i
and a Grandstream BT-200.
Paging person dials 777 and records msg. Msg is then played to other
handsets when # is pressed. Remember, the person paging can't hangup
until the page has been played (in this example).
HTH
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish
patel
Sent: 15 March 2011 15:17
To: asterisk-users
Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom
Hey,
Could you give me some idea how to do this ? I meant record and play ?
do you want me to use .call file ?
-Satish
Date: Mon, 14 Mar 2011 16:29:19 +
From: a...@datavox.co.uk
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom
If I was worried I'd record the 'page' first - and then play it back
to
50 handsets at a time (using a loop).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish
patel
Sent: 14 March 2011 16:25
To: asterisk-users
Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom
Thanks Kevin,
I test page application and it works but i am worried about i have 200
SIP phone. Do you think asterisk page application can handle that
number
of page ?
Just worried about my asterisk. I don't want to crach :(
-Satish
Date: Mon, 14 Mar 2011 11:18:36 -0500
From: kpflem...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom
On 03/14/2011 10:01 AM, satish patel wrote:
Hey Guys,
I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi
stopped working look like asterisk 1.8 did some changes in manager
apps
i am doing following.. my phone is ringing but not auto answer
could
you
give me some issue what i am doing wrong ?
The manager interface has indeed changed between 1.2 and 1.8 (likely
it
has changed many times), and you would do yourself a world of good
to
read through the upgrade notes that came with Asterisk 1.8 to
understand
how you might need to change your scripts.
In addition, Asterisk 1.8 has a built-in Page() application you can
use
from the dialplan to achieve what it appears you were trying to
achieve
with your AGI script.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype:
kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
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Re: [asterisk-users] Asterisk 1.8 paging with ploycom
If I was worried I'd record the 'page' first - and then play it back to 50 handsets at a time (using a loop). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: 14 March 2011 16:25 To: asterisk-users Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom Thanks Kevin, I test page application and it works but i am worried about i have 200 SIP phone. Do you think asterisk page application can handle that number of page ? Just worried about my asterisk. I don't want to crach :( -Satish Date: Mon, 14 Mar 2011 11:18:36 -0500 From: kpflem...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom On 03/14/2011 10:01 AM, satish patel wrote: Hey Guys, I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi stopped working look like asterisk 1.8 did some changes in manager apps i am doing following.. my phone is ringing but not auto answer could you give me some issue what i am doing wrong ? The manager interface has indeed changed between 1.2 and 1.8 (likely it has changed many times), and you would do yourself a world of good to read through the upgrade notes that came with Asterisk 1.8 to understand how you might need to change your scripts. In addition, Asterisk 1.8 has a built-in Page() application you can use from the dialplan to achieve what it appears you were trying to achieve with your AGI script. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 paging with ploycom
...http://ofps.oreilly.com/titles/9780596517342/ch11.html if you're not sure on Multicast (near the bottom). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes Sent: 14 March 2011 16:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom On 14 Mar 2011, at 16:24, satish patel wrote: I test page application and it works but i am worried about i have 200 SIP phone. Do you think asterisk page application can handle that number of page ? Do they support multicast? S If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 paging with ploycom
Oops - from the very bottom of that page (no pun intended) : So far as we can tell, Polycom sets do not support multicast. We certainly were not able to find a way to use it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes Sent: 14 March 2011 16:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom On 14 Mar 2011, at 16:24, satish patel wrote: I test page application and it works but i am worried about i have 200 SIP phone. Do you think asterisk page application can handle that number of page ? Do they support multicast? S If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help on incoming
...or for DAHDI channnels - the same thing in chan_dahdi.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bakko Sent: 07 March 2011 19:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help on incoming Hi, for sip channels, look at faxdetect options on the sip.conf file BR - Andrea If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gosub and 'h' (again?)
Thanks for your reply - but I did it a slightly different way:
Nevermind - I've re-written my dialplan so that all subs are in one
context. Now I only need 1 more line of code.
Thanks anyway :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faisal
Hanif
Sent: 06 March 2011 01:54
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Gosub and 'h' (again?)
Well a solution for you to put original context name in variable and
then use that variable in goto statement on h.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew
Thomas
Sent: Friday, March 04, 2011 4:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Gosub and 'h' (again?)
Problem as follows:
[default]
exten = 777,1,Gosub(sub,1,1)
exten = 777,n,Hangup()
exten = h,1,NoOp(hung up in 'default' context)
[sub]
exten = 1,1,NoOp(in sub)
exten = 1,n,Playback(tt-monkeys)
exten = 1,n,Return()
exten = h,1,NoOp(hung up in 'sub' context)
This works fine if the caller listens to all the 'tt-monkeys' and let's
the system hangup. You get the hang up in the 'default' context.
But, if the caller hangs up BEFORE the end of 'tt-monkeys' - the hang up
occurs in the 'sub' context. This means that I have to force each sub
routine to go to the main contexts 'h' extension ('exten =
h,1,Goto(default,h,1)' in this case).
Is there a way to tell * to use the default 'h' extension on a hang up -
rather than having to put a 'h' in to every separate sub routine?
I know Tilghman said ...Gosub, on the other hand, isn't really even
executing at that point, so there isn't a code path that exists whereby
the Gosub can empty the return stack and return to the original
place [see
http://lists.digium.com/pipermail/asterisk-dev/2008-May/033153.html].
But what does that mean in English ;)?
Thanks
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents of
this message, and any attachments, are the property of DataVox, and are
intended for the confidential use of the named recipient only. If you
are not the intended recipient, employee or agent responsible for
delivery of this message to the intended recipient, take note that any
dissemination, distribution or copying of this communication and its
attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable. Every effort has been made
to ensure that this e-mail or any attachments are free from viruses.
While the company has taken every reasonable precaution to minimise this
risk, neither company, nor the sender can accept liability for any
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you should carry out your own virus checks before opening any
attachments.
Registered in England. No. 27459085.
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Re: [asterisk-users] mySQL connection testing
Danny - Thanks, but that wouldn't work either - as I am fetching
multiple rows (not in that example - but I do in a production
environment).
Steve - If mySQL in the dialplan is so bad - why did Digium include it
in the first place? JFYI - I use mySQL in the dialplan all the time -
and it always works a treat - first time, every time. I do use AGI for
'other' things (eg. I've completely re-written the AgentCallbackLogin
feature in php) and that also works a treat. Each to their own I guess.
Anyway - back to the question (repeated in case it got lost amongst all
this) Is there a way to check if a specific MYSQL connection id is
connected or not?.
BTW - using a 'disconnect {connid}' twice doesn't actually break
anything - it just causes an error on the console. So I can live with a
'no' answer.
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
Sent: 03 March 2011 17:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] mySQL connection testing
On Thu, 3 Mar 2011, Andrew Thomas wrote:
Gentlemen, can we please not turn this in to an Asterisk and DB
commands
bashing thread?
I'm just suggesting that maybe you are 'swimming upstream' trying to use
MySQL within the dialplan.
Much the same as if you were proposing an office system using a 'tin
cans
and string' mesh with carrier pigeons for out of band call signaling and
having a problem with poop buildup on the endpoints -- I might propose
using Asterisk :)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
PST
Newline Fax:
+1-760-731-3000
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Re: [asterisk-users] mySQL connection testing
Thanks Tilghman - this is exactly what I wanted to hear. As for the 'inclusion' bit - true, but it's still infused in to the addons package at the Digium end (isn't it?). Anyway, I'll go create a mysql.conf file now :) Cheers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: 04 March 2011 08:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mySQL connection testing On Thursday 03 March 2011 08:42:42 Andrew Thomas wrote: Does anybody know of a way to test whether a mySQL connection invoked from the dialplan is current or not? There is no way to test it. If you want this, you should track the information yourself or don't disconnect anywhere but in the h extension. BTW, the disconnect is not strictly needed in all versions of the addons since 1.4.9. Due to the possibility of a memory leak, the connections are tracked and deleted when the channel is destroyed. See this issue (and the patch) for more information: https://issues.asterisk.org/view.php?id=14757 -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Gosub and 'h' (again?)
Problem as follows:
[default]
exten = 777,1,Gosub(sub,1,1)
exten = 777,n,Hangup()
exten = h,1,NoOp(hung up in 'default' context)
[sub]
exten = 1,1,NoOp(in sub)
exten = 1,n,Playback(tt-monkeys)
exten = 1,n,Return()
exten = h,1,NoOp(hung up in 'sub' context)
This works fine if the caller listens to all the 'tt-monkeys' and let's
the system hangup. You get the hang up in the 'default' context.
But, if the caller hangs up BEFORE the end of 'tt-monkeys' - the hang up
occurs in the 'sub' context. This means that I have to force each sub
routine to go to the main contexts 'h' extension ('exten =
h,1,Goto(default,h,1)' in this case).
Is there a way to tell * to use the default 'h' extension on a hang up -
rather than having to put a 'h' in to every separate sub routine?
I know Tilghman said ...Gosub, on the other hand, isn't really even
executing at that point, so there isn't a code path that exists whereby
the Gosub can empty the return stack and return to the original
place [see
http://lists.digium.com/pipermail/asterisk-dev/2008-May/033153.html].
But what does that mean in English ;)?
Thanks
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
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Re: [asterisk-users] Gosub and 'h' (again?)
Nevermind - I've re-written my dialplan so that all subs are in one
context. Now I only need 1 more line of code.
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew
Thomas
Sent: 04 March 2011 11:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Gosub and 'h' (again?)
Problem as follows:
[default]
exten = 777,1,Gosub(sub,1,1)
exten = 777,n,Hangup()
exten = h,1,NoOp(hung up in 'default' context)
[sub]
exten = 1,1,NoOp(in sub)
exten = 1,n,Playback(tt-monkeys)
exten = 1,n,Return()
exten = h,1,NoOp(hung up in 'sub' context)
This works fine if the caller listens to all the 'tt-monkeys' and let's
the system hangup. You get the hang up in the 'default' context.
But, if the caller hangs up BEFORE the end of 'tt-monkeys' - the hang up
occurs in the 'sub' context. This means that I have to force each sub
routine to go to the main contexts 'h' extension ('exten =
h,1,Goto(default,h,1)' in this case).
Is there a way to tell * to use the default 'h' extension on a hang up -
rather than having to put a 'h' in to every separate sub routine?
I know Tilghman said ...Gosub, on the other hand, isn't really even
executing at that point, so there isn't a code path that exists whereby
the Gosub can empty the return stack and return to the original
place [see
http://lists.digium.com/pipermail/asterisk-dev/2008-May/033153.html].
But what does that mean in English ;)?
Thanks
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents of
this message, and any attachments, are the property of DataVox, and are
intended for the confidential use of the named recipient only. If you
are not the intended recipient, employee or agent responsible for
delivery of this message to the intended recipient, take note that any
dissemination, distribution or copying of this communication and its
attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable. Every effort has been made
to ensure that this e-mail or any attachments are free from viruses.
While the company has taken every reasonable precaution to minimise this
risk, neither company, nor the sender can accept liability for any
damage which you sustain as a result of viruses. It is recommended that
you should carry out your own virus checks before opening any
attachments.
Registered in England. No. 27459085.
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[asterisk-users] mySQL connection testing
Does anybody know of a way to test whether a mySQL connection invoked
from the dialplan is current or not?
For example:
extensions.conf
===
[context]
exten = _X.,1,MYSQL(Connect connid localhost user pass db)
exten = _X.,n,MYSQL(Query resultid ${connid} SELECT `something` FROM
`table` WHERE `number` = ${EXTEN})
exten = _X.,n,MYSQL(Fetch foundRow ${resultid} something)
exten = _X.,n,MYSQL(Clear ${resultid})
exten = _X.,n,Wait(10) ; just for fun
exten = _X.,n,MYSQL(Disconnect ${connid})
exten = _X.,n,Hangup()
exten = h,1,MYSQL(Disconnect ${connid})
Now if the caller hangs up before the 10 second timeout - then all is
well. But, if they don't, Asterisk tries to disconnect an already
disconnected connection. I need a way of detecting that the connection
has already been disconnected - so I don't try and disconnect it again.
Something like: exten = h,1,ExecIf(CHECK IF CONNECTION STILL OPEN - IN
CASE CALLER HUNG UP AFTER TIME-OUT)?MYSQL(Disconnect ${connid}))
Any ideas?
Ta
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Re: [asterisk-users] mySQL connection testing
The wait is there as a test. This gives the 'tester' the option of
hanging up before the disconnect or not.
Either way - a connection can still be left open if the caller hangs up
before the first disconnect.
This is my problem. See the 'h' line.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: 03 March 2011 14:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] mySQL connection testing
Andrew Thomas wrote:
exten = _X.,n,MYSQL(Clear ${resultid})
exten = _X.,n,Wait(10) ; just for fun
exten = _X.,n,MYSQL(Disconnect ${connid})
You should be doing the wait after the disconnect.
Doug
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Re: [asterisk-users] mySQL connection testing
MYSQL_STATUS???
Is this documented anywhere (as I can't seem to find anything about this
variable)?
Remember, I need to test whether a specific {connid} is still connected
or not - not the whole mySQL connection.
As for the 'wait' command in my example - it is there purely for testing
purposes - otherwise you'd have to be damn quick to beat the disconnect
(but it's not impossible)!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: 03 March 2011 15:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] mySQL connection testing
Andrew Thomas wrote:
The wait is there as a test. This gives the 'tester' the option of
hanging up before the disconnect or not.
And the purpose for that would be to share available connections?
I've always considered it bad to leave a connection open and have always
closed them down after a query.
Either way, I test for mysql connection errors by:
exten = s,n,GotoIf($[${MYSQL_STATUS} = -1]?mysql_failed,s,1)
Doug
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Re: [asterisk-users] mySQL connection testing
I found that after I typed :) Trouble is - that variable gets triggered after every MYSQL command - not just the disconnect one. So it's no good to me I'm afraid. Thanks for trying. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: 03 March 2011 16:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mySQL connection testing Andrew Thomas wrote: MYSQL_STATUS??? Is this documented anywhere (as I can't seem to find anything about this variable)? core show application mysql hylafax*CLI -= Info about application 'MYSQL' =- [Synopsis] Do several mySQLy things [Description] MYSQL(): Do several mySQLy things Syntax: MySQL. On exit, always returns 0. Sets MYSQL_STATUS to 0 on success and -1 on error. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mySQL connection testing
That's your opinion - and your entitled to it sir. However, this still doesn't answer my question. Cheers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: 03 March 2011 16:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mySQL connection testing On Thu, 3 Mar 2011, Andrew Thomas wrote: Does anybody know of a way to test whether a mySQL connection invoked from the dialplan is current or not? I've never been a fan of using database commands in the dialplan. I prefer to wrap up all the database cruft into a nice little black box, an AGI, where I have full access to the database API and real debugging tools. I think database commands in the dialplan are just ugly. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mySQL connection testing
Gentlemen, can we please not turn this in to an Asterisk and DB commands bashing thread? All I want is a simple answer to a simple question - not a debate on using AGI/AMI or any other methods. Thanks for your co-operation. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: 03 March 2011 16:32 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mySQL connection testing Danny Nicholas wrote: Not to mention that Asterisk is developmental and a moving target. And that's why I'm still on 1.4. And, I have no experience with AGI, nor have I had the time to tackle it in the last 6 months. When I finally do move over to 1.8 series, I plan on looking into ODBC. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failover Routing
It seems like it is a v1.8 only function at present (unless a backport is
released).
From http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
-
Asterisk 1.8 will allow to read SIP response codes in the dialplan via
${HASH(SIP_CAUSE,channel-name)}
Asterisk 1.8 also comes with a 'use_q850_reason' configuration option for
generating and parsing, if available:
-
That will give you what you want if you consider upgrading to v1.8.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: 01 March 2011 16:24
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Failover Routing
Try this - it says it is for 1.8 but might work in 1.6
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepika Nijhawan
Sent: Tuesday, March 01, 2011 10:18 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Failover Routing
SIP_HEADER() gives you only access to headers of the initial INVITE request
(and not, for example, the final BYE message) How will I check sip response
with this like 404 or 503?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: 01 March 2011 15:09
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Failover Routing
-Original Message-
From: Bob Beers [mailto:bob.be...@gmail.com]
Sent: 01 March 2011 13:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Deepika Nijhawan
Subject: Re: [asterisk-users] Failover Routing
On Tue, Mar 1, 2011 at 8:09 AM, Deepika Nijhawan deepika.nijha...@oxygen8.com
wrote:
Ya, below is my routing:
Exten = 1234,1,Dial(SIP/abc)
Exten = 1234,n,Dial(SIP/xyz)
If 1234 is unallocated on SIP/abc it returns 1 in ${HANGUPCAUSE}
variable. For this I don't want it to try SIP/xyz. But overall, if we
get SIP 4xx reason then call should hangup like it
sends
back 404 not found for this case and if we get SIP 5xx response then
should
try SIP/xyz.
Is there any way to check sip responses and do failover routing based
on that?
Have you looked at SIP_HEADER() dialplan function?
https://wiki.asterisk.org/wiki/display/AST/Function_SIP_HEADER
Maybe you can parse Reason header in 4xx or 5xx response?
HTH,
-Bob
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepika Nijhawan
Sent: Tuesday, March 01, 2011 9:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Failover Routing
It says it for asterisk1.8. I am using asterisk1.6, can we use this function in
this version. Is it possible for you to give example on how to use?
I just went into my 1.4.37 console and find that SIP_HEADER is there in Core
show functions so it should be useable in 1.6.
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Re: [asterisk-users] [OT] Yealink IP Phones
It's all I use now. I was luckily enough to be involved with quite a bit of the beta testing in the UK - and, although there are a couple of 'nice-to-haves' missing, they are excellent handsets. Polycom sound quality at Grandstream prices ;) I particularly like the 'use your own screen logo' option. A gimmick maybe - but a nice one! Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]-- Sent: 25 February 2011 17:04 To: asterisk-users@lists.digium.com Subject: [asterisk-users] [OT] Yealink IP Phones Hello all, After numerous issues with Snom phones (360/370/870) potentially looking to migrate too Yealink as their product range looks very promising indeed. Are any of you using them with Asterisk ? How do they perform ? Do you use mass deployment at all ? Would be very interested to hear from you. -- Thanks, Phil If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4] Comparing value of string with spaces?
Changing
exten = start,n,While($[${MYVAR} != Some string])
to
exten = start,n,While($[${MYVAR} != Some string])
does the trick for me.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: 02 March 2011 13:25
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] [1.4] Comparing value of string with spaces?
Hello
I haven't found an example on how to compare the value of a
string variable with spaces in it, and the While loop below never exits:
== extensions.conf
exten = start,n,Set(MYVAR=Dummy value)
exten = start,n,NoOp(${MYVAR})
;BAD TOO
;exten = start,n,While(!$[${MYVAR} : Some string])
exten = start,n,While($[${MYVAR} != Some string])
exten = start,n,Set(MYVAR=Some string)
exten = start,n,EndWhile()
==
Thank you.
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Re: [asterisk-users] cmd MySQL
Try rrplacing MySQL(Query resultid ${conn_id} SELECT/ ramal/ FROM/
colaboradores/ WHERE/ ramal=${EXTEN});
With MySQL(Query resultid ${conn_id} SELECT `ramal` FROM
`colaboradores` WHERE `ramal`='${EXTEN}');
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felipe
Figueiredo
Sent: 18 February 2011 17:57
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] cmd MySQL
Hi guys,
I'm trying to connect Asterisk to the MySQL, but I can't execute it. It
returns an error, as below:
-- Executing [200@teste:2] MYSQL(Console/dsp, Query resultid 1
SELECT/ ramal/ FROM/ colaboradores/ WHERE/ ramal=200) in new stack
[Feb 18 15:55:13] WARNING[7696]: app_mysql.c:393 aMYSQL_query:
aMYSQL_query: mysql_query failed. Error: You have an error in your SQL
syntax; check the manual that corresponds to your MySQL server version
for the right syntax to use near '/ ramal/ FROM/ colaboradores/ WHERE/
ramal=200' at line 1
Its seems it can connect to mysql
My extension (AEL) is:
MySQL(Connect conn_id localhost root 123456 crm);
MySQL(Query resultid ${conn_id} SELECT/ ramal/ FROM/
colaboradores/ WHERE/ ramal=${EXTEN});
MySQL(Fetch fetchid ${resultid} RAMAL);
MySQL(Clear ${fetchid});
MySQL(Disconnect ${connid});
MySQL(Clear ${connid});
NoOp(${RAMAL});
Where is the error? Thanks!!
The MySQL server is in the same server where Asterisk is running.
Thanks!!!
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Re: [asterisk-users] calls are not going thru e1 line
This is very strange. Everything matches mine except Asterisk itself
(I'm using 1.6.2.16.1).
I did notice that you set the loadzone(s) for UK use - yet your e-mail
address in in Poland. Are you setting this up in the UK?
BTW - you have a typo in chan_dahdi.conf (busydetec=yes is missing the
't' [I wonder if this is causing your problem - as the 'include' is
after this]) and I'd cetainly remove pulsedial=yes ;).
Anyway, here's the part of my chan_dahdi.conf that is working for me
(I've changed the context to match yours):
;chan_dahdi.conf
[trunkgroups]
[channels]
language = en
context = incoming_calls
switchtype = euroisdn
pridialplan = unknown
prilocaldialplan = unknown
internationalprefix = 00
nationalprefix = 0
localprefix =
unknownprefix =
rxwink = 300
usecallerid = yes
hidecallerid = no
callwaiting = yes
usecallingpres = yes
sendcalleridafter = 1
callwaitingcallerid = yes
threewaycalling = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
rxgain = 0.0
txgain = 0.0
group = 1
callgroup = 1
pickupgroup = 1
immediate = no
faxdetect = no
echocancel = yes
echocancelwhenbridged = no
echotraining = yes
signalling = pri_cpe
channel = 1-15,17-31
Maybe drop mine in as a replacement and see what happens then (remember
to back yours up).
BTW - you don't need to include dahdi-channels.conf in the above - as
it's already included.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Albert
Sent: 21 February 2011 13:53
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] calls are not going thru e1 line
Hi Andrew,
I am using current versions of software, find below versions:
1.) asterisk
voice:~# asterisk -V
Asterisk 1.8.2.3
2.)dahdi
*CLI dahdi show version
DAHDI Version: 2.4.0 Echo Canceller: MG2
3.) lipri
*CLI pri show version
libpri version: 1.4.11.5
I've already tried to call over each channel from 1 to 15 (i have only
15B channels)
exten = _X.,n,Dial(DAHDI/1/${EXTEN})
exten = _X.,n,Dial(DAHDI/2/${EXTEN})
exten = _X.,n,Dial(DAHDI/15/${EXTEN})
but everytime i am getting the same DIALSTATUS
snip
-- Channel 0/1, span 1 got hangup request, cause 31
...
-- Auto fallthrough, channel 'SIP/2000-0002' status is 'CHANUNAVAIL'
/snip
Regards,
Robert
On 21.02.2011 12:13, Andrew Thomas wrote:
I'm curious as to what versions of everything you are using. Reason
being this line -- DAHDI/i1/00256312261627-1 is proceeding passing
it to SIP/5000-.
It states DAHDI/i1/00256312261627-1... and I don't recall seeing that
before (my 2.4.0 says -- DAHDI/1-1 is proceeding passing it to
SIP/801-000c [1-1 being the span and channel numbers]).
What happens if you change exten = _X.,n,Dial(DAHDI/g1/${EXTEN}) to
exten = _X.,n,Dial(DAHDI/1/${EXTEN})?
If you have received this communication in error we would appreciate
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and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
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Re: [asterisk-users] calls are not going thru e1 line
I'm curious as to what versions of everything you are using. Reason
being this line -- DAHDI/i1/00256312261627-1 is proceeding passing
it to SIP/5000-.
It states DAHDI/i1/00256312261627-1... and I don't recall seeing that
before (my 2.4.0 says -- DAHDI/1-1 is proceeding passing it to
SIP/801-000c [1-1 being the span and channel numbers]).
What happens if you change exten = _X.,n,Dial(DAHDI/g1/${EXTEN}) to
exten = _X.,n,Dial(DAHDI/1/${EXTEN})?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Albert
Sent: 17 February 2011 16:56
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] calls are not going thru e1 line
On 17.02.2011 17:47, Danny Nicholas wrote:
snip
Post your dahdi show channels output.
Have you checked the lines with a regular handset?
here it is:
*CLI dahdi show status
Description Alarms IRQbpviol CRC4
Fra Codi Options LBO
T2XXP (PCI) Card 0 Span 1OK 0 0 0
CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
T2XXP (PCI) Card 0 Span 2UNCONFI 0 0 0
CAS Unk 0 db (CSU)/0-133 feet (DSX-1)
*CLI dahdi show channels
Chan Extension Context Language MOH Interpret
BlockedState
pseudodefaultdefault
In Service
1incoming_calls en default
In Service
2incoming_calls en default
In Service
3incoming_calls en default
In Service
4incoming_calls en default
In Service
5incoming_calls en default
In Service
6incoming_calls en default
In Service
7incoming_calls en default
In Service
8incoming_calls en default
In Service
9incoming_calls en default
In Service
10incoming_calls en default
In Service
11incoming_calls en default
In Service
12incoming_calls en default
In Service
13incoming_calls en default
In Service
14incoming_calls en default
In Service
15incoming_calls en default
In Service
17incoming_calls en default
In Service
18incoming_calls en default
In Service
19incoming_calls en default
In Service
20incoming_calls en default
In Service
21incoming_calls en default
In Service
22incoming_calls en default
In Service
23incoming_calls en default
In Service
24incoming_calls en default
In Service
25incoming_calls en default
In Service
26incoming_calls en default
In Service
27incoming_calls en default
In Service
28incoming_calls en default
In Service
29incoming_calls en default
In Service
30incoming_calls en default
In Service
31incoming_calls en default
In Service
Yes, is was checked and calls were going through line.
Regards,
Albert
If you have received this communication in error we would appreciate
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If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
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Every effort has been made to ensure that this e-mail or any attachments
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accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
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Re: [asterisk-users] Question about EuroBRI final 2 digits
This sounds like a job for DISA. http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA to see if DISA helps. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: 03 February 2011 19:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Question about EuroBRI final 2 digits Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call before they finish dialing all 8 digits. Is there a way to prevent this, or to catch the additional 2 digits somewhere in the stream? The receptionist is unhappy because she gets all the 6-digit calls and must then transfer. Is this a p2p vs p2mp issue? Thanks in advance, Cassius Smith If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about EuroBRI final 2 digits
...or there :) Anyway AT sends the call before they finish dialling all 8 digits means that they don't send all the digits. Conflicting sentence in OP. Perhaps it would help if the OP could determine if AT actually send 6 or 8 digits in the signalling (I reckon it's 6). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Beers Sent: 10 February 2011 14:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about EuroBRI final 2 digits On Thu, Feb 10, 2011 at 6:08 AM, Andrew Thomas a...@datavox.co.uk wrote: This sounds like a job for DISA. http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA to see if DISA helps. If OP is using Asterisk18, perhaps we should direct him to look here? https://wiki.asterisk.org/wiki/display/AST/Application_DISA cheers, -- -Bob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to insert cdr-data into mysql-DB
Try changing 'hostname=127.0.0.1' to 'hostname=localhost' in the cdr_mysql.conf. I seem to remember a problem I had when '127.0.0.1' and 'localhost' didn't marry up never did find out why. If that doesn't work - try GRANT SELECT , INSERT ,UPDATE ON `Asterisk`.`cdr` TO 'asteriskcdr'@'localhost'; instead. HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: 24 January 2011 15:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Unable to insert cdr-data into mysql-DB Hello list, I keep on getting the error : ERROR[1707] cdr_addon_mysql.c: Cannot connect to database server 127.0.0.1: (1045) Access denied for user 'asteriskcdr'@'localhost' (using password: YES) I have a 'cdr' table in my MySQL-DB. On this table the user 'asteriskcdr' has select, insert, update privileges. GRANT SELECT , INSERT ,UPDATE ON `Asterisk`.`cdr` TO 'asteriskcdr'@'127.0.0.1'; cdr_mysql.conf : [global] hostname=127.0.0.1 dbname=Asterisk table=cdr password=mysecret user=asteriskcdr port=3306 sock=/tmp/mysql.sock userfield=1 I really don't know why Asterisk cannot connect to the table.. Kind regards, Jonas. If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MOH and parking
Hi Leif, Submitted as requested - https://issues.asterisk.org/view.php?id=18672 Thanks -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen Sent: 21 January 2011 15:58 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] MOH and parking On 11-01-21 08:52 AM, Andrew Thomas wrote: I know that the 'fix' has just been applied (https://issues.asterisk.org/view.php?id=18262) - but why does it stop the moh only to start it again? This, also, seems to cause a CDR problem (see below). After speaking with Shaun and Russell, this is likely related to some other part of code, and the fix that went in shouldn't have caused this issue. It's possible fixing this may have caused some other part of the code that was broken to be more prevalent though. Could you open an issue on bug tracker? Thanks! Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mailing list question
Thank you Kevin. That's exactly the answer I was after. I'll see if I can get it 'stopped' at our server end. BTW - the reason I asked in here was so that everyone could see the answer and, hopefully, do the same. Thanks again! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: 20 January 2011 18:44 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mailing list question On 01/20/2011 11:16 AM, Andrew Thomas wrote: Sorry Dannny - it didn't work :( I can only hope that someone at API has the answer. Thanks for trying :) API provides the physical services and bandwidth for the mailing lists, but does not operate them. If you go to the lists.digium.com site and choose the 'asterisk-users' mailing list, you can see there is a link to send a message to the list administrator(s)... which would probably be more effective than asking a question like this on the list itself :-) In any case, the answer is no... the lists are operated using Mailman software, and it essentially leaves the message bodies alone (although it does do scrubbing of attachments in some cases). Unless you want to include your signature as an attachment marked as something other than 'text', I don't believe there's any way to get the mailing list process to drop your signature block. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH and parking
I know that the 'fix' has just been applied (https://issues.asterisk.org/view.php?id=18262) - but why does it stop the moh only to start it again? This, also, seems to cause a CDR problem (see below). -- Executing [7000@chambers:1] Park(SIP/2000-0008, ) in new stack == Parked SIP/2000-0008 on 7001 (lot default). Will timeout back to extension [chambers] s, 1 in 60 seconds -- Added extension '7001' priority 1 to parkedcalls (0xb6fd1160) -- SIP/2000-0008 Playing 'digits/7.gsm' (language 'en') -- SIP/2000-0008 Playing 'digits/0.gsm' (language 'en') -- SIP/2000-0008 Playing 'digits/0.gsm' (language 'en') -- SIP/2000-0008 Playing 'digits/1.gsm' (language 'en') == Spawn extension (chambers, s, 1) exited non-zero on 'Parked/SIP/2000-0008ZOMBIE' -- Stopped music on hold on DAHDI/1-1 -- Started music on hold, class 'dv-ip', on DAHDI/1-1 [Jan 21 13:39:17] ERROR[22913]: cdr_addon_mysql.c:313 mysql_log: Failed to insert into database: (1062) Duplicate entry 'DV-IP-1295617064.8' for key 1 == Spawn extension (park-dial, SIP02000, 1) exited non-zero on 'SIP/2000-0008ZOMBIE' BTW 'DV-IP-1295617064.8' is the CDR entry for when the call first came in to the queue. So, it looks like it's trying to use the same unique ID again. If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Accessing a 'user' variable via. dialplan.
Hi,
I know you can access various sip variables via
'Set(sstatus=${SIPPEER(201,status)})' (for example) to get the status of
the sip user - but what about variables?
I have a user that has setvar=123456 in their users.conf (sip.conf if
you prefer). I can read it with a 'sip show peer 201' - but that gives
everything and parsing that isn't really an option.
Anyone know how to access 'variables' (and maybe the contents) directly?
Thanks
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
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Re: [asterisk-users] Accessing a 'user' variable via. dialplan.
That's what I am already using :)
Somehow, the outbound ID sometimes gets messed up (maybe to do with 2
calls from different users at once) - and the wrong one is sent to the
telco.
So, rather than just using a 'Set(CALLERID(num)=callidnum' just before
Dial - I wanted to check the user directly (to double-check Asterisk if
you like and check my own sanity).
Something alone the lines of 'Set(idvar=${SIPPEER(201:callidnum)})' or
even 'Set(idvar=${SIPPEER(201:variables)})' [to parse that little bit
myself]. That way I can check if there is a genuine problem - or if,
indeed, it is the telco themselves (I don't want to leave a Trend tester
on-site).
Thanks anyway.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell
Sent: 20 January 2011 16:31
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Accessing a 'user' variable via. dialplan.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Andrew Thomas
Sent: Thursday, January 20, 2011 11:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Accessing a 'user' variable via. dialplan.
Hi,
I know you can access various sip variables via
'Set(sstatus=${SIPPEER(201,status)})' (for example) to get the status
of the sip user - but what about variables?
I have a user that has setvar=123456 in their users.conf (sip.conf if
you prefer). I can read it with a 'sip show peer 201' - but that
gives everything and parsing that isn't really an option.
Anyone know how to access 'variables' (and maybe the contents)
directly?
Thanks
Posted by Joshua Colp dated 12/19/2010, with the subject of Specifying
DID for outbound calls
I'm surprised nobody has suggested using the setvar functionality. It's
extremely useful for stuff like this and would allow you to keep all
CallerID information with the actual configuration of the device.
Using a configuration entry for sip.conf in another response as an
example:
[101]
type=friend
username=101
secret=
mailbox=101
callerid=User One 101
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes
setvar=EXTERNAL_CALLERID=User One 3012323434
And then in extensions.conf:
exten = _1NXXNXX,1,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten = _1NXXNXX,n,Dial(SIP/${EXTEN}@vitel-outbound)
Of course you could add some sanity checking there to make sure that
${EXTERNAL_CALLERID} contains a value and if not default to your main
DID.
-
I think you can get an idea on how to access setvar much easier, he also
stated you can have multiple setvar(s)
Ie,
Setvar=VAR_1=Taco
Setvar=VAR_2=Apples
Setvar=VAR_3=Bannanna
--
William Stillwell
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If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
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Re: [asterisk-users] context problem
I always thought the last bit (after the /) is where the context in
sip.conf landed.
What about:
(sip.conf)
register = 119909:pas...@sip.prov.org/52525252
register = 119909:pas...@sip.prov.org/59595959
[52525252]
...
context = TRUNKin52
...
[59595959]
...
context = TRUNKin59
...
And split them out in extensions.conf?
I have a suspicion that you have 'context=TRUNKin' under the '[default]'
section of sip.conf - which is why they are hitting there in the first
place.
Then again, I have been known to be wrong ;)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 20 January 2011 16:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] context problem
On 01/20/2011 05:23 PM, Jeroen Eeuwes wrote:
Hi Jonas,
What else can I try ?
Yeah, Asterisk always assumes that from 1 ip address there can only be
inbound number. Not very user-friendly.
I think I've used something like this:
exten = s,1,Set(CALL-TO=${SIP_HEADER(TO)})
exten = s,n,Set(CALL-FROM=${CALLERIDNUM})
exten = s,n,GotoIf($[${CALL-TO} :
.*52525252.*]?TRUNKin,52525252,1)
exten = s,n,GotoIf($[${CALL-TO} :
.*59595959.*]?TRUNKin,59595959,1)
exten = s,n,etcetera
Best regards,
Jeroen Eeuwes
--
Hello,
this is the result when using your config :
[Jan 20 17:33:50] -- Executing [s@TRUNKin:1]
NoOp(SIP/119909-06d7, context TRUNKin - s) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:2]
NoOp(SIP/119909-06d7, 775006 775006) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:3]
NoOp(SIP/119909-06d7, 775006 775006) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:4]
NoOp(SIP/119909-06d7, sip:s@11.11.12.112) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:5]
NoOp(SIP/119909-06d7, ) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:6]
NoOp(SIP/119909-06d7, 775006) in new stack
dialplan :
exten = s,1,NoOp(context TRUNKin - s)
exten = s,n,NoOp(${CALLERID(all)})
exten = s,n,NoOp(${CALLERID(all)})
exten = s,n,NoOp(${SIP_HEADER(TO)})
exten = s,n,NoOp(${CALLERIDNUM})
exten = s,n,NoOp(${CALLERID(num)})
Kind regards,
Jonas.
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of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
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criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
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accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
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[asterisk-users] Mailing list question
Hi, Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? In other words - something like disclaimer at the end of my message would inform the list software to remove any lines after it. My massive disclaimer is added by the server you see - and it's now annoying me - let alone the rest of the list. Ta If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mailing list question
Let's see :) -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 20 January 2011 17:04 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Mailing list question -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Thomas Sent: Thursday, January 20, 2011 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Mailing list question Hi, Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? In other words - something like disclaimer at the end of my message would inform the list software to remove any lines after it. My massive disclaimer is added by the server you see - and it's now annoying me - let alone the rest of the list. Putting the -- in front of it might make it go away. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mailing list question
That's my last option Jon. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder Sent: 20 January 2011 16:59 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mailing list question On 01/20/2011 12:01 PM, Andrew Thomas wrote: why not just subscribe with an account that doesn't do that like gmail or yahoo ? Hi, Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? In other words - something likedisclaimer at the end of my message would inform the list software to remove any lines after it. My massive disclaimer is added by the server you see - and it's now annoying me - let alone the rest of the list. Ta If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mailing list question 2
Sorry about this - testing this disclaimer problem :) -- If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mailing list question
Sorry Dannny - it didn't work :( I can only hope that someone at API has the answer. Thanks for trying :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 20 January 2011 17:04 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Mailing list question -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Thomas Sent: Thursday, January 20, 2011 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Mailing list question Hi, Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? In other words - something like disclaimer at the end of my message would inform the list software to remove any lines after it. My massive disclaimer is added by the server you see - and it's now annoying me - let alone the rest of the list. Putting the -- in front of it might make it go away. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mailing list question 2
Tell you what Steve - I'll not take you up on your kind offer - I'll just let my server keep adding the disclaimer. There - problem solved. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 20 January 2011 17:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Mailing list question 2 On 20 Jan 2011, at 17:13, Andrew Thomas wrote: Sorry about this - testing this disclaimer problem :) I can give you a POP3 account on my server if it stops you spamming the list?.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
Top posting? Who cares? Get a life! Now - can we get back to Asterisk et al? Thanks! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Murawski Sent: 18 January 2011 02:57 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Top Posting On 01/17/2011 08:26 PM, Matt Riddell wrote: On 17/01/11 4:29 PM, jon pounder wrote: Surely there is some mail client smart enough to be able to flip around the levels of indenting so most recent is top or bottom. If not quit bitching and make one - I will continue top posting since I don't seem to be alone in preferring it. That was one of the first things that came to mind. I'm definitely more keen on inline replies - if you reply to 20 points in someone's email you quote the part you're replying to then reply to it. That was the standard for much of the 90's for emails. I do like that method but most people don't seem to do it anymore. In a long email it's the only way. Otherwise you'd scroll down to find the question, scroll up to find the answer, scroll down to find the next question, scroll up for the next answer etc - crazy. It's also easier to keep the context of what's going on. If replying in one big block, I try to keep the style of one paragraph of response for each paragraph of question, but sometimes stuff just mixes in between and you can easily lose context. Much easier when replies are inline with the questions. It gets hard to follow when there's a dozen nested levels of reply. In conclusion, I think it just depends (tm). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sound quality issue
Something that often gets forgotten is the on-site LAN infrastructure as well. It could be a bad/faulty switch, rubbish cabling, induced interference etc. etc. all at the customers premises. Maybe a handset plugged directly in to the back of the router, before it hits the LAN would tell you whether the call is actually getting 'distorted' en-route or not? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: 16 January 2011 12:28 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sound quality issue Le 15/01/2011 20:38, Cédric Lemarchand a écrit : Hello, Hi [...] I am sure there are RTP packets losses somewhere, except RTP debug in the asterisk CLI, how can i determine where the problem come from ? [...] You don't tell which protocol (SIP, IAX, H323) nor which asterisk version. FYI Asterisk 1.6.2.15 in iax had audio quality problems, solved in 1.6.2.16. If you have the possibility, connect directly a phone to the server, eg Device - LAN (no MPLS) - Asterisk. Check also if a call to echo test has the same bad quality. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
Why do I top post? Simple. I read every message in the thread - and if there are 10 messages (for example) in that thread - then why should I have to read them all over again on the last one? Top posting is here - to stay! Stop being so anal and 'retro'. Bottom posting belongs in forums - top post belongs in e-mail lists. There - said it! As for my sig/disclaimer - how about 10 copies of it before you get a reply? That's what bottom posting would have done for you! Anyway Digium, Inc. | Software Developer means you should be developing software - not replying to inane posts like mine :P Have a nice day! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: 18 January 2011 14:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Top Posting On 11-01-18 04:22 AM, Andrew Thomas wrote: Top posting? Who cares? Get a life! Clearly not you, so why both even replying? At worst case it is just redundant information for people, best case somebody reads the email thread at starts bottom posting. I suggest taking a moment and re-reading the thread. If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. Additionally, do you really need a 17 line[1] signature? [1] - http://s3.amazonaws.com/theoatmeal-img/comics/email/4.png -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
I also agree this is a pointless discussion because, clearly, nobody is willing to budge, and it has nothing to do with Asterisk. Amen :) [oh no, a bottom post] If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
SEE THE BOTTOM :P -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: 18 January 2011 16:18 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Top Posting On Tue, Jan 18, 2011 at 03:18:49PM -, Andrew Thomas wrote: Why do I top post? Simple. I read every message in the thread - and if there are 10 messages (for example) in that thread - then why should I have to read them all over again on the last one? You mean: why should I have to read 10 messages worth of lines just to figure what you're talking about? It is interesting to note that your mailer (MS-Outlook) has very bad support for threading. In fact, it (combined with the MS-Exchange server) does not really bother reproducing the mail headers that are required to keep the proper threading. Which is why you get a big pile of messages and have to resort to keeping everything in the message itself. Top posting is here - to stay! Top posted content has just been cut off :-) not replying to inane posts like mine :P So, you really want this thread to go on forever? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You mean: why should I have to read 10 messages worth of lines just to figure what you're talking about? Nope! I mean: why should I have to read the SAME 10 messages worth of lines over and over... It is interesting to note that your mailer (MS-Outlook) has very bad support for threading. In fact, it (combined with the MS-Exchange server) does not really bother reproducing the mail headers that are required to keep the proper threading. Oh dear God! You mean I'm using a Micro$oft product(s)? I'll go shoot myself now! Well, after I've shot every other M$ user! Top posted content has just been cut off :-) I chuckled :-) So, you really want this thread to go on forever? Yeah! I'm having bit of a slow CBA day at work... Watch out - here comes that damned disclaimer again: If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Push central phone book to phones
For the Yealink - you can use a 'remote' XML file. The XML is stored on a web server and is retrieved by the phone every time you press the phones 'key'. This has the advantage of not needing the directory to be pushed to the handset - and the handset always gets the latest version. Of course, the XML file needs to be kept up to date every time someone's name/extn changes. HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: 03 December 2010 13:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Push central phone book to phones On 12/02/2010 04:31 PM, Ishfaq Malik wrote: On Thu, 2010-12-02 at 16:01 +0100, Jonas Kellens wrote: On 12/02/2010 03:47 PM, Ishfaq Malik wrote: On Thu, 2010-12-02 at 15:19 +0100, Jonas Kellens wrote: Hello, I have Snom, Cisco, Grandstream YeaLink phones. Is there a way to push a centralized phone book to these phones ?? Kind regards, Jonas. -- With Snom phones (and also Yealink I think) you can use centralised LDAP directories on a server This is a public server on the internet. I don't think I can use LDAP to push then ? Kind regards, Jonas. If you can set up and administer LDAP on the server you will be able to use it on the Snom (and maybe Yealink) phones. I can use different Organizational Units for different phone books ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to hangup all channels
2 ways:
Read http://www.voip-info.org/wiki/view/Asterisk+AGI
or in PHP - system (asterisk -rx 'core restart now' /dev/null);
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giuseppe D'alessio
Sent: 29 November 2010 14:47
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to hangup all channels
Thank you, i want to follow your idea, how i can send and receive data from/to
Command Line in PHP Script?
Thank you in advance
Date: Sat, 27 Nov 2010 08:45:47 -0800
From: asterisk@sedwards.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to hangup all channels
On Sat, 27 Nov 2010, Giuseppe D'alessio wrote:
Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels.
1) sudo /etc/init.d/asterisk restart
2) Write a script to do asterisk -r -x 'core show channels', parse the
output and do asterisk -r -x 'channel request hangup ${CHANNEL_NAME} for
each channel.
3) Write a script to do #2 using AMI.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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its attachments is strictly prohibited, and may be subject to civil or
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Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
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Re: [asterisk-users] How to hangup all channels
Re-top-posting... I was merely pointing out that AGI exists (teach a man to fish...)! Sorry for not being as perfect as you... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: 29 November 2010 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to hangup all channels Un-top-posting... From: Giuseppe D'alessio Thank you, i want to follow your idea, how i can send and receive data from/to Command Line in PHP Script? On Mon, 29 Nov 2010, Andrew Thomas wrote: Read http://www.voip-info.org/wiki/view/Asterisk+AGI An AGI is executed in the context of a channel. Are you suggesting the OP write an AGI so he can call into his system to ask it to hang up all channels? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test BRI lines energy saving mode ?
The D-channel isn't actually 'dropped' - it is put in to a 'power-save' state. See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to 'Activation / Deactivation' for more information. Anyway - this is a known 'problem' - https://issues.asterisk.org/view.php?id=17270 As there is no fix for the above - then I doubt * will be able to emulate the NT's function. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: 07 October 2010 01:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to test BRI lines energy saving mode ? Olivier wrote: Hello, If my understanding is correct, these days it seems that many ISDN BRI lines are configured in energy saving mode in which signalling D-channel is dropped until a new call comes in. Is it possible to replicate this behaviour with Asterisk (when Asterisk is in NT mode and is seen as a public ISDN by another PBX, for instance) ? If not, would you it would be a useful addition to Asterisk ? Regards Energy saving??? I don't think so. If the D channel is down, how would I make an outgoing phone call? Something in this mode or your explanation just does not sound right... Lyle Giese LCR Computer Services, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test BRI lines energy saving mode ?
Well, to go slightly O/T: If you read the issue tracker for 17270 - it appears to be a LibPri 'fault'. So I would say that the main work would need to be in LibPri Q:is this how DAHDI talks to the ISDN?. Maybe someone who knows LibPri and DAHDI better can explain how the two combine... Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 07 October 2010 11:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to test BRI lines energy saving mode ? 2010/10/7 Andrew Thomas a...@datavox.co.uk The D-channel isn't actually 'dropped' - it is put in to a 'power-save' state. See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to 'Activation / Deactivation' for more information. Anyway - this is a known 'problem' - https://issues.asterisk.org/view.php?id=17270 As there is no fix for the above - then I doubt * will be able to emulate the NT's function. Thanks for these interesting links ! So this Activation/Desactivation feature seems to be missing in Asterisk. Would you say it should be implemented in libpri, in dahdi, or both ? If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] debian/dahdi/zaphfc - Unable to receive TEI fromnetwork!
What happens if you change to: signalling=bri_cpe_ptp -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Sent: 01 October 2010 11:37 To: asterisk-users@lists.digium.com Subject: [asterisk-users] debian/dahdi/zaphfc - Unable to receive TEI fromnetwork! Hello, snip # cat /etc/asterisk/chan_dahdi.conf [trunkgroups] [channels] language=fr switchtype=euroisdn ... group=1 signalling=bri_cpe_ptmp context=from-isdn channel = 1-2 snip If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Go from *100* to just 100
${EXTEN:1:3}
http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/
asterisk-CHP-5-SECT-3.html#asterisk-CHP-5-SECT-3.6.3
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 30 September 2010 08:31
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Go from *100* to just 100
Hello list,
how can I go from *100* to 100 ?
I know I can do something like ${EXTEN:1} but that way I only get rid of
just one *.
Kind regards,
Jonas.
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
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Re: [asterisk-users] Weird Behavior with DAHDI
Downgrade your LibPri instead (1.4.10.2 is fine). See https://issues.asterisk.org/view.php?id=17270 for more info. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias Sent: 29 September 2010 13:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Weird Behavior with DAHDI Hello, I'm experiencing some weird problems on my server: - 1) The following messages are filling up my logs: [Sep 29 08:24:59] WARNING[7077]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! [Sep 29 08:24:59] WARNING[7078]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 171 as D-channel anyway! [Sep 29 08:24:59] WARNING[7075]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 78 as D-channel anyway! [Sep 29 08:24:59] WARNING[7079]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 202 as D-channel anyway! [Sep 29 08:24:59] WARNING[7073]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! [Sep 29 08:24:59] WARNING[7080]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 233 as D-channel anyway! [Sep 29 08:24:59] WARNING[7074]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 47 as D-channel anyway! In the Asterisk CLI, i'm watching these messages constantly 2) I've plugged in a real E1 PRI ISDN: r...@sangoma-testing:/usr/src# asterisk -rx 'pri show spans' PRI span 1/0: Provisioned, In Alarm, Down, Active PRI span 2/0: Provisioned, In Alarm, Down, Active PRI span 3/0: Provisioned, In Alarm, Down, Active PRI span 4/0: Provisioned, Up, Active PRI span 5/0: Provisioned, In Alarm, Down, Active PRI span 6/0: Provisioned, In Alarm, Down, Active PRI span 7/0: Provisioned, In Alarm, Down, Active PRI span 8/0: Provisioned, In Alarm, Down, Active Seems to be OK! but i can't make a call: -- Executing [691918...@pbx1:1] Dial(SIP/xtravoip200-021a47e0, DAHDI/g4/691918892|30|m) in new stack [Sep 29 08:29:51] WARNING[7338]: channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 [Sep 29 08:29:51] WARNING[7338]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [691918...@pbx1:2] Hangup(SIP/xtravoip200-021a47e0, ) in new stack == Spawn extension (pbx1, 691918892, 2) exited non-zero on 'SIP/xtravoip200-021a47e0' What is happening? Could you let me know how to debug or to understand the output from pri intense debug span 4? TEI: 0 State 7(Multi-frame established) V(A)=30, V(S)=30, V(R)=30 K=7, RC=0, l3initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ 00 01 01 3d ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 030 P/F: 1 0 bytes of data -- Starting T200 timer Sangoma-Testing*CLI TEI: 0 State 8(Timer recovery) V(A)=30, V(S)=30, V(R)=30 K=7, RC=0, l3initiated=0, reject_except=0, ack_pend=0 T200_id=1, N200=3, T203_id=0 [ 02 01 01 3d ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 030 P/F: 1 0 bytes of data TEI: 0 State 8(Timer recovery) V(A)=30, V(S)=30, V(R)=30 K=7, RC=0, l3initiated=0, reject_except=0, ack_pend=0 T200_id=1, N200=3, T203_id=0 [ 02 01 01 3d ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 030 P/F: 1 0 bytes of data -- Got ACK for N(S)=30 to (but not including) N(S)=30 Done handling message for SAPI/TEI=0/0 Sangoma-Testing*CLI TEI: 0 State 8(Timer recovery) V(A)=30, V(S)=30, V(R)=30 K=7, RC=0, l3initiated=0, reject_except=0, ack_pend=0 T200_id=1, N200=3, T203_id=0 [ 00 01 01 3d ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 030 P/F: 1 0 bytes of data -- Got ACK for N(S)=30 to (but not including) N(S)=30 -- Stopping T200 timer -- Starting T203 timer What should i check on the above span debug? what's important there? the timers? the ACK? SAPI? TEI? is there any place to learn how to understand this output? Hope you can help me Verions of my server: libpri version: 1.4.11.4 Asterisk 1.4.24.1 DAHDI Version: 2.4.0 WANPIPE Release: 3.5.15.4 Please don't ask me to upgrade my Asterisk Version, the idea is to test this environment Best Regards! If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential
Re: [asterisk-users] DAHDI FXO port only recognizes the S extension?
The cause is bad programming. You can't go from an 's' to an '_X.' the way you tried. exten =s,1,Answer() exten =s,n,Wait(1) exten =s,n,Dial(DAHDI/3) exten =s,n,Hangup Is correct (that's why it works). What is it you are trying to achieve? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Songtao Yu Sent: 29 September 2010 10:56 To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI FXO port only recognizes the S extension? Hi All, When I tried to write my dial plan as below for my FXO port, which connects one PSTN line: [from-pstn] exten =s,1,Answer() exten =s,n,Wait(1) exten =_X.,1,Dial(DAHDI/1) exten =_X.,n,Hangup I got the following message: Connected to Asterisk 1.6.2.13 currently running on fax (pid = 8154) Verbosity was 0 and is now 4 -- Starting simple switch on 'DAHDI/1-1' -- Executing [...@from-pstn:1] Answer(DAHDI/1-1, ) in new stack -- Executing [...@from-pstn:2] Wait(DAHDI/1-1, 1) in new stack -- Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN' -- Hungup 'DAHDI/1-1' But if I changed the _X. to S extension, I can get the whole thing to work well: [from-pstn] exten =s,1,Answer() exten =s,n,Wait(1) exten =s,n,Dial(DAHDI/3) exten =s,n,Hangup Would you please let me which casuses this issue? Thanks, Songtao Yu If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not able to join conference
I was wondering what happened if YOU put that number in. Does it put everyone in to the same conference? That would, at least, prove that the MeetMe app was working as it should (unless you've tried this already). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: 20 September 2010 14:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Not able to join conference it's going to put you in conf no 500 without prompting you to enter a conference number I guess, but i don't it's going to solve my issue. actually I'm atill wondering is there a way to debug just Meetme app output or the only way is turn the whole debug thing on? On Mon, Sep 20, 2010 at 4:11 AM, Andrew Thomas a...@datavox.co.uk wrote: What happens if you put in a 'room' number? Eg: exten = 8080,3,MeetMe(500|MDci) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: 17 September 2010 14:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Not able to join conference Hi All, We are running to a weird problem, we're using asterisk 1.2 as a production server (I'm wiling to move very soon to more recent version) and our problem is when somebody try to join a conference he's told that he's the only one in the conference but in fact there is some 3 or 5 or whatever people in that same conference, after several tries he can/cannot enter the conference and meet with the people already in, here is the lines corresponding to conf in the dialplan, that would be a big help if you guys can help diagnose the issue. exten = 8080,1,Answer exten = 8080,2,Wait,1 exten = 8080,3,MeetMe(|MDci) If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Abdullah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not able to join conference
What happens if you put in a 'room' number? Eg: exten = 8080,3,MeetMe(500|MDci) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid touati Sent: 17 September 2010 14:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Not able to join conference Hi All, We are running to a weird problem, we're using asterisk 1.2 as a production server (I'm wiling to move very soon to more recent version) and our problem is when somebody try to join a conference he's told that he's the only one in the conference but in fact there is some 3 or 5 or whatever people in that same conference, after several tries he can/cannot enter the conference and meet with the people already in, here is the lines corresponding to conf in the dialplan, that would be a big help if you guys can help diagnose the issue. exten = 8080,1,Answer exten = 8080,2,Wait,1 exten = 8080,3,MeetMe(|MDci) If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime semi-colon
I'd forgot about doing it that way (I use that for $).
Thanks for the memory jog :)
Cheers
Andy
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 16 September 2010 13:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime semi-colon
On 16 Sep 2010, at 12:56, Andrew Thomas wrote:
Does anyone know how to send * a semi-colon from a realtime database.
I know that * uses the semi-colon as a 'seperator' - but I need to be
able to use one in a command. I know I can use \; in the non-realtime
configs, but this doesn't work in realtime.
in /etc/asterisk/extensions.conf
[globals]
SEMICOLON=\;
Then use ${SEMICOLON} in realitime Hacky, but it's what I'm using at
the moment..
S
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Every effort has been made to ensure that this e-mail or any attachments
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before opening any attachments.
Registered in England. No. 27459085.
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[asterisk-users] Realtime semi-colon
Hi list, Does anyone know how to send * a semi-colon from a realtime database. I know that * uses the semi-colon as a 'seperator' - but I need to be able to use one in a command. I know I can use \; in the non-realtime configs, but this doesn't work in realtime. Cheers, Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtimemysql
This is a problem with extconfig.conf - not your res_ or cdr_ ones. In your case - extconfig.conf probably contained something like 'sippeers = mysql,MyDBase,sippeers'. The 'problem' is that the middle parameter is no longer for the database name - it is for the context in res_mysql.conf. So, the above now becomes 'sippeers = mysql,general,sippeers'. Give that a go... Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: 08 September 2010 15:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtimemysql Hello, in asterisk 1.4.30 all realtime configurations go well. In asterisk 1.6.2.11 the following appears on CLI : [Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql: MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf) [Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql: MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf) res_mysql.conf : [general] dbhost = 127.0.0.1 dbname = MyDBase dbuser = asterisk dbpass = mysecret dbport = 3306 dbsock = /tmp/mysql.sock requirements=warn ; or createclose or createchar What do I need to change to be conform asterisk 1.6 ?! Reloading, restarting asterisk and restarting my CentOS-server all doesn't help. Jonas. If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High volume BLF - Suggestions?
As a side note to this - do NOT try and use Aastra's - as they tend to crash after 50 BLF's! Also, could you please send me (perhaps off-list to a...@datavox.co.uk) your Yealink T28 findings - as I am a beta tester for them? Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 13 September 2010 11:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] High volume BLF - Suggestions? 2010/9/13 Steve Davies davies...@gmail.com Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite low-end hardware, but we cannot find any handsets that can cope with it longer term. Our test involves about 10 BLF-NOTIFY messages per second to each handset with a 5-second pause every 5 seconds. This will either crash or render unusable all of the following combinations: snom360 + 1 x sidecar As Snom phones have a parameter to express a time period during which BLF's SUBSCRIBE messages are spread and sent to Asterisk, I thought Snom phones would handle this load more easily. Yealink T28 + 1 x sidecar Yealink T28 + 2 x sidecar Cisco SPA504g + 1 x sidecar Cisco SPA504g + 2 x sidecar Cisco SPA525g + 1 x sidecar (reboots often) Cisco SPA525g + 2 x sidecar (reboots quickly) Aastra 55i + non-LCD sidecar Did not try Polycom as they do not do directed pickup and only small sidecars. Linksys SPA962 with one sidecar is OK but is discontinued hardware. Help? Thanks, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'dirty' upgrade of 1.4
Thanks to everyone who replied. This is great news ;). I'll get the thing upgraded tonight (when it's quiet). Thanks again. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner Sent: 26 July 2010 16:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 'dirty' upgrade of 1.4 When you run make, it compiles the binaries in the src directory. Once it is done compiling stop asterisk. Running make install will copy the compiled binaries into their respective folders on your system. Then just start asterisk. If you need to revert, stop asterisk, run make install in the old src directory, then start asterisk. Ryan On Mon, Jul 26, 2010 at 9:45 AM, Andrew Thomas a...@datavox.co.uk wrote: Hi Danny, I understand (and welcome) the separate src directories. This would allow me to 'revert' should I feel the need (assuming I can just re-compile over each one). I just need to know if I can re-compile over the existing first. Thanks for your reply. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 26 July 2010 14:15 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 'dirty' upgrade of 1.4 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- Subject: [asterisk-users] 'dirty' upgrade of 1.4 Apologies if this has been asked before. Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1? Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the source files for 1.4.34 over the top of the existing 1.4.24.1 files. Also, will I need to stop * to perform this routine - or can I just 'upgrade' and then do a * 'restart'? Question 1 - unless you are un-tarring to a specific directory, you would have /usr/local/src/asterisk-1.4.24.1 and /usr/local/src/asterisk-1.4.34 segregated source trees. Question 2 - you don't have to stop asterisk, but you should (best practice?) since installing a new release usually involves removing/replacing the .so files in /usr/lib/asterisk/modules. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 'dirty' upgrade of 1.4
Apologies if this has been asked before. Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1? Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the source files for 1.4.34 over the top of the existing 1.4.24.1 files. Obviously, I will need to keep my config files (and sound files etc) - so I'll back them up first. Also, will I need to stop * to perform this routine - or can I just 'upgrade' and then do a * 'restart'? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'dirty' upgrade of 1.4
Hi Danny, I understand (and welcome) the separate src directories. This would allow me to 'revert' should I feel the need (assuming I can just re-compile over each one). I just need to know if I can re-compile over the existing first. Thanks for your reply. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: 26 July 2010 14:15 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] 'dirty' upgrade of 1.4 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- Subject: [asterisk-users] 'dirty' upgrade of 1.4 Apologies if this has been asked before. Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1? Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the source files for 1.4.34 over the top of the existing 1.4.24.1 files. Also, will I need to stop * to perform this routine - or can I just 'upgrade' and then do a * 'restart'? Question 1 - unless you are un-tarring to a specific directory, you would have /usr/local/src/asterisk-1.4.24.1 and /usr/local/src/asterisk-1.4.34 segregated source trees. Question 2 - you don't have to stop asterisk, but you should (best practice?) since installing a new release usually involves removing/replacing the .so files in /usr/lib/asterisk/modules. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Aastra 50-limit blf
Hello all, Just wondering if anyone ever solved the Aastra 50-BLF limit when used with Asterisk (any flavour)? I know it's not strictly and Asterisk question - but I'm sure there's plenty of you out there using Aastra's on the end. Cheers, Andrew Thomas dCAP #1473 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing for incoming call
exten = did,1,Answer exten = did,n,Playtones(ring) exten = did,n,Wait(10) exten = did,n,StopPlaytones() exten = did,n,BackGround(sound file) did = the DID number as presented and note the '1' before Answer. This works for me. exten = 820055,1,Answer() exten = 820055,n,PlayTones(ring) exten = 820055,n,Wait(5) exten = 820055,n,StopPlayTones() exten = 820055,n,[do something interesting from now on] That's my DID (820055) being answered first and then waiting for 5 seconds. I use it for fax detect this way. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Smither Sent: 18 December 2009 23:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ringing for incoming call Dear All, I am using Asterisk 1.4 on CentOS 5. I have an incoming DID provided by Vitelity. When the number is called it goes to my Asterisk box. The protocol is SIP. This all works just fine if I answer the call and begin a playback. I want to let the number ring for a few seconds before it is answered, and would like the caller to hear it ringing. I have tried: ... exten = s,n,Answer exten = s,n,Playtones(ring) exten = s,n,Wait(10) exten = s,n,StopPlaytones() exten = s,n,BackGround(sound file) ... also ... exten = s,n,Answer exten = s,n,Ringing() exten = s,n,Wait(10) exten = s,n,BackGround(sound file) ... I have also tried moving the Answer app to right before the BackGround app. In all cases when I call the number I never hear it ringing. After the 10 second delay, the BackGround app does run. Connecting to the CLI does not give me any useful information - for example the Ringing app is shown to run, but the caller does not hear it. Any suggestions? Many thanks! -- Bob Smither smit...@c-c-i.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stutter playback
This sounds more like the alarm system putting pulses/tones on the line (maybe the alarm has a dialler/anti-cut-line-detection? So, as the alarm is adding stuff AFTER the asterisk box - I doubt you will see anything on the PC itself. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Samad Sent: 22 August 2009 04:48 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] stutter playback On Fri, Aug 21, 2009 at 08:53:23AM -0400, Steve Totaro wrote: On Fri, Aug 21, 2009 at 8:39 AM, Alex Samad a...@samad.com.au wrote: Hi I had a working system, until recently - its asterisk 1.6.1 from debian - not the lastest as the last doesn't seem to work. but somebody who rang me said my voice mail announcement was all stuttery. so i dialed my voicemail box and its really stuttery... so I have done a reboot and its just as bad, now I am not sure what to check to try and get this working again . Alex I would check cpu, diskpace, memory, I/O, network wasn't that, I have a alarm system on the backup pstn line, seems like there is something wrong there, cause when I remove the alarm system from the equation everything seems okay, so I am guessing it was causing some problem on my tdm410 card. strange thing is i did not see any spikes on io , cpu, network... Alex -- Think of it! With VLSI we can pack 100 ENIACs in 1 sq. cm.! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent Agent Login from a second extension
The only way around the 'auto-logout' problem I found was to call a script when agents login. This script checks to see if they are already logged in or not - then, if they are, it does whatever I want (I manually log them off the other phone first - you could play a message instead). HTH Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shanavaz E A Sent: 02 September 2009 07:27 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Prevent Agent Login from a second extension Hi friends, Is there any way to prevent an Agent from logging in from a second extension if he is already logged on from an extension. Right now, the scenario is if he login from a second extension, asterisk will automatically log him off from first extension. What I need is that asterisk should tell him that he is already logged on from an extension and should prevent him from logging in again from another extn. The problem with existing scenario is that, I am not getting CDR record for the automatic log out event. I need this for evaluation purposes. I am using asterisk 1.2.30. I have 1.4 also but that also is having the same behavior. Thanks in advance for any help. Regards Shanavaz. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] onnecting two asterisk using B410p BRI cards
...and did you switch the termination dip switches over (on the NT ports of the B410P card)? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of voip crazy Sent: 17 August 2009 07:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] onnecting two asterisk using B410p BRI cards I just plug the junper in NT mode with no success. VoipCrazy 2009/8/15 Paul Hales pdha...@optusnet.com.au: Use a standard network cable - but you have to activate the 'terminate' jumper on the NT end. - Also, the new BRI stuff in dahdi is much easier to work with than misdn. PaulH voip crazy wrote: Hello all, I'm trying to conect two asterisk servers using two B410p Digium cards. One card on each server. I just setting up the first BRI port on server A as nt_ptp and the first BRI port on server B as te_ptp. I use an ethernet wire to connect the first port of server A (nt_ptp) with the first port on server B (te_ptp) but the port light cotinues blinking on red on both sides once the cable was pluged. Then I use an isdn crossover wire with this king of schema and the lights get blinking red again. Tx+ 3 --+ +- 3 . X Rx+ 4 --+ +- 4 . Tx- 5 --+ +--5 . X Rx- 6 --+ +--6 In both servers when I do in asterisk CLI misdn shos stacks, the port one on each machine shows Server A: BEGIN STACK_LIST: * Port 1 Type TE Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 Server B: BEGIN STACK_LIST: * Port 1 Type NT Prot. PTP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 Which kind of cable should I use? Why both in ports L1Link is failed? How could I solve that? Any clue will be welcomed. Thanks in advance. VoipCrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Underscore won't help as that's for pattern matching. Under the sip conf, have you tried adding 'fromuser=8001187e0' to the [8001187e0] bit? I have this in my Sipgate setup and it works. Worth a try. Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Plattes Sent: 10 August 2009 11:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] context does not work Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
V1.6.1.0 [9290740] type = peer username = 9290740 fromuser = 9290740 secret = you-wish! host = sipgate.co.uk fromdomain = sipgate.co.uk insecure = port,invite context = inbound caninvite = no canreinvite = no nat = yes disallow = all allow = ulaw allow = alaw dtmfmode = info qualify = 5000 That works for me. Any inbound call to my 9290740 number goes to my inbound context and does what it should. PS - Don't forget to do a 'sip reload' when you change the sip.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Plattes Sent: 10 August 2009 13:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] context does not work Hi Andrew, it didn't help. Which version of Asterisk do you use? Thanks On Mon, Aug 10, 2009 at 1:55 PM, Andrew Thomasa...@datavox.co.uk wrote: Underscore won't help as that's for pattern matching. Under the sip conf, have you tried adding 'fromuser=8001187e0' to the [8001187e0] bit? I have this in my Sipgate setup and it works. Worth a try. Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Plattes Sent: 10 August 2009 11:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] context does not work Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possibly I don't understand sip peers
[peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 read what you've put!!! The 'allow' should be 'permit' as Jared already told you (and he should know what he's talking about). insecure=port,invite -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Ferrell Sent: 29 July 2009 23:34 To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Possibly I don't understand sip peers Jared Smith wrote: On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote: I have a carrier who tells me he will be sending me traffic from a wide range of IP addresses. so I set up a realtime peer as follows: [peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 insecure=port,invite Yes, he's really claiming to originate from any of the IP in the block When I leave the host blank, we reject calls with a 404. shouldn't I be able to put in a kind of wildcard for his IP block or am I just being silly? If not, what am I doing wrong? I think you've got your syntax wrong there... permit and deny statements are used to create Access Control Lists and to limit the IP address ranges. The allow and disallow statements are to allow or disallow various codecs. They way you've specified it above, you're allowing a codec called xxx.xxx.xxx.0/255.255.255.0, which probably isn't what you want. I have the codec permissions in the columns allow and disallow. Those seem to work ok. it's permit/deny/mask I seem to be having a problem with. Like I say, I don't think I understand their use or perhaps they don't work in realtime ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold based on user
You can use 'Set(CHANNEL(musicclass)=${MOH})' anywhere in your dialplan - so
you can set it at any stage of an inbound or outbound call (as long as it is
before the Dial/Queue command).
Eg:
[inbound]
exten = _X.,1,Set(CHANNEL(musicclass)=${MOH})
exten = _X.,n,Dial(whomever-you-want)
[outbound]
exten = _X.,1,Set(CHANNEL(musicclass)=${MOH})
exten = _X.,n,Dial(where-ever-you-want)
Then, when 'whomever-you-want' puts the call on hold - they get
'whomever-you-want's MOH.
Simples :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen
Sent: 24 July 2009 14:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Music on hold based on user
Andrew Thomas schrieb:
I do this using the setvar facility in sip.conf.
eg. setvar=MOH=music1
Then in the dialplan (extensions.conf) all you need to do is
'Set(CHANNEL(musicclass)=${MOH})'
Juan C. Crespo R. wrote:
Guys I wonder if its possible to set a different MoH based on
groups, I mean if one of the Admin group put on hold the call play
music 1, if another from Technical Support put on hold the call play
music 3, something like this
Admin - Music1
Contrallors - Music 2
Technical Support - Music 3
The way I understood the OP was that he wants different MoH classes
depending on the callee (not depending on the caller).
Philipp Kempgen
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