Re: [asterisk-users] Realtime SIP peer status
On 04/13/2011 09:18 PM, Rob Coward wrote: Rather than add extra overhead to your dialplan and the asterisk server, why not make use of the AMI and have a background process listening for the various events and updating your database accordingly ? See http://www.voip-info.org/wiki/view/asterisk+manager+events#ExtensionStatusEvent and http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.5/lib/Asterisk/AMI.pm#Events Regards, Rob Hello, this event tells me something about an extension, but not about the SIP peer status. 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
[asterisk-users] sip error logging
I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. the last entry I have is: /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: Registration from 'sip:22942@10.0.0.3' failed for '10.0.0.228:5060' - No matching peer found my logger.conf looks like: # grep -v '^;' /etc/asterisk/logger.conf [general] [logfiles] console = notice,warning,error,dtmf messages = notice,warning,error,verbose,dtmf,fax if i send 'options' or 'register' from a non-configured sip peer, i dont see anything in the log. am I missing something ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] sip error logging
On 4/15/2011 3:39 AM, Jeremy Kister wrote: I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. forgot to mention that I can replicate this behavior on 1.8.2.3 and 1.8.3.2 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] Existing Asterisk 1.8 upgrade with new release
On 110414 1346, satish patel wrote: We are running asterisk 1.8.3.2 and if in future i need to upgrade with new release then how should i upgrade it. The reason i am asking is i did lots of customization in make menuselect so do i need to keep all option remember or is there anyway i just copy paste existing configuration file in new tarball and just run make make install something like that... How you guys doing this process ? Correct, keep your menuselect.makeopts file. -kkm -- _ -- 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] Would a job posting be ok for this list?
We are hiring a VoIP developer. Would it be within the list guidelines to post a position description? Not sure if that's in the scope of this list, but does not match asterisk-biz either. -kkm -- _ -- 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
Registry type Event will give you information about your peer. Adolphe Cher-aime From my Iphone On Apr 15, 2011, at 1:15 AM, Jonas Kellens jonas.kell...@telenet.be wrote: On 04/13/2011 09:18 PM, Rob Coward wrote: Rather than add extra overhead to your dialplan and the asterisk server, why not make use of the AMI and have a background process listening for the various events and updating your database accordingly ? See http://www.voip-info.org/wiki/view/asterisk+manager+events#ExtensionStatusEvent and http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.5/lib/Asterisk/AMI.pm#Events Regards, Rob Hello, this event tells me something about an extension, but not about the SIP peer status. 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 -- _ -- 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
On 04/15/2011 11:53 AM, Adolphe Cher-aime wrote: Registry type Event will give you information about your peer. Adolphe Cher-aime From my Iphone I don't find information on how this event tells me whether the SIP peer is occupied with a call or not. How can I capture the notify messages (as in BLF) ?? 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
[asterisk-users] How to get back park call
Hi ,
How I will get my park call back?
*1 ) features.conf *
[general]
parkext = 900; What extension to dial to park
parkpos = 901-920 ; What extensions to park calls on.
context = bhati-park
*2) Extensions.conf *
[incoming]
include = bhati-park
exten = XXX,1,Answer
exten = XXX,n,Dial(SIP/${EXTEN},30,tTr)
exten = XXX,n,Hangup()
When I park the call with dial #900 then Asterisk give me park extension
901. and call parked but then I want to get back park call then asterisk
read my incoming context not bhati-park context.
please tell me how I get back park call before specify time in features.conf
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
--
_
-- 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] [OT] 802.11x roaming
Hi, I'd like to replace my DECT + fxs phones with some wireless phones. Main problem is: what about the roaming from one ap to another? Do someone adopted the 802.11r standard? What APs and what phones do you suggest me? I'm open to suggestions but I'd like to avoid proprietary solutions, I don't want to be bound to any hardware vendor. My switch does support spanning tree protocol. Thank you, Darkbasic -- _ -- 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] Friday April 15 at 12 Noon EDT
Hi all, You're welcome as always to join the talk on the VoIP Users Conference, VUC for short. VUC began as the Asterisk Users Conference but for obvious reasons, we changed the name in the first year, although Digium was our sponsor for three years. We still have plenty of you who are asterisk users and developers on this our fifth year. Come on by, listen, talk ro text on IRC. All the links are on the main URL: http://vuc.me Today's guest is Matt Bramson of InPhonex to talk about the launch of Televate. The second hour we talk about anything you can think of. If you have a g722-capable SIP client or phone, call 200...@login.zipdx.com and get on #vuc on irc.freenode.net We're also testing a bridge from Gtalk so you can do both voice and IRC with that today, by adding voipusersconfere...@gmail.com to your contact list and calling it. Thanks to asterisk community and VUC member Tim Panton for that effort and for skype:vuc.me See you there in a little over 4 hours, :r -- _ -- 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] Friday April 15 at 12 Noon EDT
Hi I want to join this conference but please tell me the topic of conference and the process of joining step by step. Please tell me the time in IST too On Fri, Apr 15, 2011 at 5:09 PM, randulo rand...@randulo.com wrote: Hi all, You're welcome as always to join the talk on the VoIP Users Conference, VUC for short. VUC began as the Asterisk Users Conference but for obvious reasons, we changed the name in the first year, although Digium was our sponsor for three years. We still have plenty of you who are asterisk users and developers on this our fifth year. Come on by, listen, talk ro text on IRC. All the links are on the main URL: http://vuc.me Today's guest is Matt Bramson of InPhonex to talk about the launch of Televate. The second hour we talk about anything you can think of. If you have a g722-capable SIP client or phone, call 200...@login.zipdx.com and get on #vuc on irc.freenode.net We're also testing a bridge from Gtalk so you can do both voice and IRC with that today, by adding voipusersconfere...@gmail.com to your contact list and calling it. Thanks to asterisk community and VUC member Tim Panton for that effort and for skype:vuc.me See you there in a little over 4 hours, :r -- _ -- 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 -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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] Possible bug in Hangup() (Asterisk 1.4.x)
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,Hangup(${HANGUPCAUSE})
I have noticed that no matter what value we set in the Hangup(cause
code) commands, if the call is not answered by peer01 for any reason,
the actual cause code returned to the calling party is a 503, no matter
what the ${HANGUPCAUSE} is.
Even if we set a fixed value like Hangup(1) (which should give a 404) or
Hangup(17) (which should give a 486), the cause code returned is always
a 503.
Has anyone else noticed this? I went through the issue tracker but I
couldn't find any relevant bug posted in the past. I am certain that in
previous versions I could set the reply message to the desired value, so
I wonder if this is a bug in this particular version (1.4.33.1).
--
Best regards,
Vlasis Hatzistavrou.
--
_
-- 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] Friday April 15 at 12 Noon EDT
On Fri, Apr 15, 2011 at 1:56 PM, virendra bhati virbh...@gmail.com wrote: I want to join this conference but please tell me the topic of conference and the process of joining step by step. You're welcome to join us! All this information is at the top of the main site: http://vuc.me Please tell me the time in IST too http://vuc.me/next -- _ -- 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] If voice mail not found dialplan
Hey guys, I have stdexten macro dialplan and I have to handle those who doesn't have voicemail box setup. Right now if someone call and if person unavailable the it's just hangup that call. I want it say person doest have vm setup yet. smthing like that. How should I handle this in my dialplan ? -- Sent from my iPhone -- _ -- 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] Friday April 15 at 12 Noon EDT
Is this online conf? Or are there archived files we can review? -- Sent from my iPhone On Apr 15, 2011, at 8:06 AM, randulo rand...@randulo.com wrote: On Fri, Apr 15, 2011 at 1:56 PM, virendra bhati virbh...@gmail.com wrote: I want to join this conference but please tell me the topic of conference and the process of joining step by step. You're welcome to join us! All this information is at the top of the main site: http://vuc.me Please tell me the time in IST too http://vuc.me/next -- _ -- 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] Friday April 15 at 12 Noon EDT
On Fri, Apr 15, 2011 at 2:16 PM, Satish Patel satish...@hotmail.com wrote: Is this online conf? Or are there archived files we can review? There are over 300 recordings here: http://vuc.me :r -- _ -- 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] If voice mail not found dialplan
Satish Patel wrote:
want it say person doest have vm setup yet. smthing like that. How
should I handle this in my dialplan ?
Here is a snippet of what I do:
[s-NOANSWER]
exten = s,1,Gosub(mailbox_exist,s,1)
[mailbox_exist]
exten = s,1,Set(_direct_vm=${ARG1})
exten = s,n,MailboxExists(${direct_vm}@sip)
exten = s,n,Goto(s-${VMBOXEXISTSSTATUS},1)
exten = s-FAILED,1,Answer()
exten = s-FAILED,n,Wait(1)
exten = s-FAILED,n,Playback(vm-theperson)
exten = s-FAILED,n,SayDigits(${direct_vm})
exten = s-FAILED,n,Playback(vm-nobox)
exten = s-FAILED,n,Set(CONNECTEDLINE(all)=Operator 4100)
exten = s-FAILED,n,Set(CALLERID(name)=No Mailbox)
exten = s-FAILED,n,Playback(pbx-transfer)
exten = s-FAILED,n,Goto(incoming,s,1)
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
[asterisk-users] Good by asterisk 1.4? Please not.
Security only fixes: 2011-04-21 So in six days, no more bugfix patches will committed into 1.4-branch :( Is a prolongation possible? Because 1.4 is so reliable now. It would be a great loss. And no, 1.8 is not (yet) a replacement. Kristijan -- _ -- 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] Good by asterisk 1.4? Please not.
On 16/04/11 12:33 AM, Kristijan Vrban wrote: Security only fixes: 2011-04-21 So in six days, no more bugfix patches will committed into 1.4-branch :( Is a prolongation possible? Because 1.4 is so reliable now. It would be a great loss. And no, 1.8 is not (yet) a replacement. It actually brings up a good point. We've just reverted a couple of installs from 1.6.2 because of deadlocks. What version should we be going to? -- 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
Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
My guess is since the call was never answered you should be looking at
${DIALSTATUS}
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 15, 2011, at 5:02 AM, Vlasis Hatzistavrou wrote:
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,Hangup(${HANGUPCAUSE})
I have noticed that no matter what value we set in the Hangup(cause code)
commands, if the call is not answered by peer01 for any reason, the actual
cause code returned to the calling party is a 503, no matter what the
${HANGUPCAUSE} is.
Even if we set a fixed value like Hangup(1) (which should give a 404) or
Hangup(17) (which should give a 486), the cause code returned is always a 503.
Has anyone else noticed this? I went through the issue tracker but I couldn't
find any relevant bug posted in the past. I am certain that in previous
versions I could set the reply message to the desired value, so I wonder if
this is a bug in this particular version (1.4.33.1).
--
Best regards,
Vlasis Hatzistavrou.
--
_
-- 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] Good by asterisk 1.4? Please not.
You know we don't have choise. I had remembered when we shifted 1.2 to first release of 1.4 and we had many issue. Same thing right now I'm dealing with 1.8 things take time to stabilized. Good luck!! -- Sent from my iPhone On Apr 15, 2011, at 8:33 AM, Kristijan Vrban vrban.l...@googlemail.com wrote: Security only fixes: 2011-04-21 So in six days, no more bugfix patches will committed into 1.4-branch :( Is a prolongation possible? Because 1.4 is so reliable now. It would be a great loss. And no, 1.8 is not (yet) a replacement. Kristijan -- _ -- 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] Require dialplan
ViciDial doesn't work that way, you have to use the agent web interface or the API to disposition a call. MATT--- On Mon, Apr 11, 2011 at 10:04 AM, mahesh katta maheshka...@flexydial.comwrote: Hi , In vicidial dialer I need small Dialplan require. when i call from hardphone , in that has 1to9 no.s i want define the dipositions like when i press the 1 it will goes NotIntrest, press 2 for NotAvailable. How can i configure for this. -- Best Regards, Mahesh Katta *BUZZ**WORKS* 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 -- _ -- 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] Possible bug in Hangup() (Asterisk 1.4.x)
Hello Jim,
Thank you for the reply.
The problem is not reading the ${HANGUPCAUSE} or the ${DIALSTATUS}. It
is that the Hangup(cause) command seems to ignore its argument and
just sends a 503 cause to the caller for all unanswered calls no matter
what...
Hangup(cause) was working as expected in previous versions and I
wonder if something was broken along the way that went by unnoticed. I
am just asking in the list in case I am missing something too obvious
before posting a bug.
--
Best regards,
Vlasis Hatzistavrou.
On 15/4/2011 4:22 μμ, Jim Dickenson wrote:
My guess is since the call was never answered you should be looking at
${DIALSTATUS}
--
_
-- 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] Possible bug in Hangup() (Asterisk 1.4.x)
If what you showed is your whole dialplan then none of the i or t or h
extensions are going to be executed for a non answered call.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 15, 2011, at 6:46 AM, Vlasis Hatzistavrou wrote:
Hello Jim,
Thank you for the reply.
The problem is not reading the ${HANGUPCAUSE} or the ${DIALSTATUS}. It is
that the Hangup(cause) command seems to ignore its argument and just sends
a 503 cause to the caller for all unanswered calls no matter what...
Hangup(cause) was working as expected in previous versions and I wonder if
something was broken along the way that went by unnoticed. I am just asking
in the list in case I am missing something too obvious before posting a bug.
--
Best regards,
Vlasis Hatzistavrou.
On 15/4/2011 4:22 μμ, Jim Dickenson wrote:
My guess is since the call was never answered you should be looking at
${DIALSTATUS}
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
On 15 April 2011 13:02, Vlasis Hatzistavrou vh...@kinetix.gr wrote:
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,Hangup(${HANGUPCAUSE})
I have noticed that no matter what value we set in the Hangup(cause code)
commands, if the call is not answered by peer01 for any reason, the actual
cause code returned to the calling party is a 503, no matter what the
${HANGUPCAUSE} is.
Even if we set a fixed value like Hangup(1) (which should give a 404) or
Hangup(17) (which should give a 486), the cause code returned is always a
503.
Has anyone else noticed this? I went through the issue tracker but I
couldn't find any relevant bug posted in the past. I am certain that in
previous versions I could set the reply message to the desired value, so I
wonder if this is a bug in this particular version (1.4.33.1).
Strictly speaking you can only Hangup (BYE) an answered and fully
established call. In SIP terms, a hangup that occurs before an answer
is a CANCEL, and I believe a CANCEL is always represented by a 503
code in chan_sip.
Regards,
Steve
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
The h extension is executed after the remote end peer01 rejects the call
with a 408. I verified it by altering the dialplan as:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,NoOp(Hangup cause is: ${HANGUPCAUSE})
exten = h,n,Hangup(${HANGUPCAUSE})
and I saw in the Asterisk CLI that the correct hangupcause is shown.
--
Best regards,
Vlasis Hatzistavrou.
On 15/4/2011 5:01 μμ, Jim Dickenson wrote:
If what you showed is your whole dialplan then none of the i or t or h
extensions are going to be executed for a non answered call.
--
_
-- 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] Possible bug in Hangup() (Asterisk 1.4.x)
Hello Steve, On 15/4/2011 5:07 μμ, Steve Davies wrote: Strictly speaking you can only Hangup (BYE) an answered and fully established call. In SIP terms, a hangup that occurs before an answer is a CANCEL, and I believe a CANCEL is always represented by a 503 code in chan_sip. Regards, Steve I see what you mean, but it is the called end (peer01) that rejects the call with a 408 message, it is not the originator that is canceling the call. The call flow is this: Caller-Asterisk-Peer01 and Asterisk receives a STATUS 408 message from Peer01 instead of an answer. Asterisk then sends a STATUS 503 to the Caller, instead of sending a STATUS 408. The question is how to copy the correct cause code from the terminating end to the originating end. I tried setting Hangup(1) to send a 404 to the called, a Hangup(17) to send a 486 to the caller and pretty much any other value in the Hangup() but Asterisk will keep on sending a 503. I don't believe that my memory fails me, I'm pretty sure I could set a desirable cause in the Hangup() command in previous versions... -- Best regards, Vlasis Hatzistavrou. -- _ -- 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] If voice mail not found dialplan
Perfect! Thanks
Date: Fri, 15 Apr 2011 08:21:48 -0400
From: supp...@drdos.info
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] If voice mail not found dialplan
Satish Patel wrote:
want it say person doest have vm setup yet. smthing like that. How
should I handle this in my dialplan ?
Here is a snippet of what I do:
[s-NOANSWER]
exten = s,1,Gosub(mailbox_exist,s,1)
[mailbox_exist]
exten = s,1,Set(_direct_vm=${ARG1})
exten = s,n,MailboxExists(${direct_vm}@sip)
exten = s,n,Goto(s-${VMBOXEXISTSSTATUS},1)
exten = s-FAILED,1,Answer()
exten = s-FAILED,n,Wait(1)
exten = s-FAILED,n,Playback(vm-theperson)
exten = s-FAILED,n,SayDigits(${direct_vm})
exten = s-FAILED,n,Playback(vm-nobox)
exten = s-FAILED,n,Set(CONNECTEDLINE(all)=Operator 4100)
exten = s-FAILED,n,Set(CALLERID(name)=No Mailbox)
exten = s-FAILED,n,Playback(pbx-transfer)
exten = s-FAILED,n,Goto(incoming,s,1)
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
[asterisk-users] sangoma card rx/tx gain level
Hey Guys! We had echo issue before so we replaced old PRI card with Sangoma A102D HWEC. Now my question is i set rx/txgain level 0.0 default do i need to touch this value or default is best. I have read on google and people say it should around 14844 on ztmonitor for rx/tx level same. I just use milliwatt and test my default 0.0 rx/tx level and it come around 4600. Do you think i need to make it around 14844 ? -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
[asterisk-users] Hot to make call parking to Mult tenant
Hi, I am working on call packing feature of asterisk. Call packing is working fine but I want to make this feature as multi tenant. exp:- *for A client* packing extension are parkext = 700 parkpos = 701-720 context = parkedcalls_A parkingtime = 45 *for B client *packing extension are parkext = 800 parkpos = 801-820 context = parkedcalls_B parkingtime = 45 Is it possible or not ? - Thanks and regards Virendra Bhati +91-9172341457 -- _ -- 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] Reach PSTN from another Asterisk
Dear, we have the following: - Asterisk A with E1 to PSTN connection. - Asterisk B with IAX trunk to Asterisk A - Outgoing routes between Asterisk A and B - Asterisk A with an outgoing route to PSTN with 9|. dial rule How can I reach the PSTN from Asterisk B through Asterisk A ??? Thanks a lot !!! Alejandro -- _ -- 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] accessing currents calls from outside asterisk
yes, ami is your unique answer. what is msisdns ? On Wed, Apr 13, 2011 at 3:18 PM, Albert alber...@wp.pl wrote: Hi, I am working on integration of 2 systems: asterisk and messaging platform. What I need is to access somehow information about current calls. Should I do it over AMI ? I need to be able to perform those 2 actions: - How can I obtain msisdns of current calls ? - How to hangup one of current calls ? Thanks for your help guys! Regards, Albert -- _ -- 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] Reach PSTN from another Asterisk
On server B use IAX2/iax-trunk-name/whatever-needs-to-be-dialed-on-A-to-make-call in the Dial command. like Dial(IAX2/sfserver1/9411212) -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 15, 2011, at 12:29 PM, Alejandro Cabrera Obed wrote: Dear, we have the following: - Asterisk A with E1 to PSTN connection. - Asterisk B with IAX trunk to Asterisk A - Outgoing routes between Asterisk A and B - Asterisk A with an outgoing route to PSTN with 9|. dial rule How can I reach the PSTN from Asterisk B through Asterisk A ??? Thanks a lot !!! Alejandro -- _ -- 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] Good by asterisk 1.4? Please not.
1.4 svn has a nasty bug in it at the moment. Would love to see that fixed ;) https://issues.asterisk.org/view.php?id=18951 Julian On 15 April 2011 14:22, Satish Patel satish...@hotmail.com wrote: You know we don't have choise. I had remembered when we shifted 1.2 to first release of 1.4 and we had many issue. Same thing right now I'm dealing with 1.8 things take time to stabilized. Good luck!! -- Sent from my iPhone On Apr 15, 2011, at 8:33 AM, Kristijan Vrban vrban.l...@googlemail.com wrote: Security only fixes: 2011-04-21 So in six days, no more bugfix patches will committed into 1.4-branch :( Is a prolongation possible? Because 1.4 is so reliable now. It would be a great loss. And no, 1.8 is not (yet) a replacement. Kristijan -- _ -- 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 -- Follow Ode To Politics by HB Tasker at http://twitter.com/HBTasker -- _ -- 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] 1.8.4-rc2: ReceiveFAX fails
On a test fax: -- Executing [s@incoming-fax:1] Set(DAHDI/4-1, FAXFILE=/var/spool/asterisk/fax/20110415_1825) in new stack -- Executing [s@incoming-fax:2] Answer(DAHDI/4-1, ) in new stack -- Executing [s@incoming-fax:3] ReceiveFAX(DAHDI/4-1, /var/spool/asterisk/fax/20110415_1825.tif) in new stack -- Channel 'DAHDI/4-1' receiving FAX '/var/spool/asterisk/fax/20110415_1825.tif' -- Channel 4 detected a CED tone towards the network. [Apr 15 18:25:52] WARNING[5600]: res_fax_spandsp.c:367 spandsp_log: WARNING T.30 ECM carrier not found [Apr 15 18:25:52] WARNING[5600]: res_fax_spandsp.c:367 spandsp_log: WARNING T.30 ECM carrier not found == Spawn extension (incoming-fax, s, 3) exited non-zero on 'DAHDI/4-1' /var/spool/asterisk/fax exists, but no file 20110415_1825.tif is created. Using spandsp-0.0.6-pre18, the Jan 22 release. Do the log notes of the CED tone or the T.30 ECM warning have anything to do with this? chan_dahdi.conf: [pstn] context=incoming-pstn-line ; Incoming calls go to [incoming-pstn-line] in extensions.conf faxdetect=incoming faxbuffers=6,full ; who knows what this does busydetect=yes dahdichan = 4 Any idea on how to debug this? sean -- _ -- 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] 1.8.4-rc2: ReceiveFAX fails
On Fri, Apr 15, 2011 at 07:00:27PM -0400, sean darcy wrote: ... Do the log notes of the CED tone or the T.30 ECM warning have anything to do with this? I can't help with this. faxbuffers=6,full ; who knows what this does But I do know that this faxbuffers policy basically is telling the kernel to allow ~120ms of audio to build up before shipping it out to the physical board. This is to work around problems where scheduling jitter on the host computer would prevent the asterisk process from keeping the DAHDI channel full and fax machines do not handle gaps in the audio very well. This is more pertinent if you're connecting a real fax machine to an FXS port. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 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
Re: [asterisk-users] 1.8.4-rc2: ReceiveFAX fails
On Fri, Apr 15, 2011 at 7:00 PM, sean darcy seandar...@gmail.com wrote: Using spandsp-0.0.6-pre18, the Jan 22 release. You might try using spandsp-0.0.6-pre17. That version works great for me with 1.8.4-rc2. When I tried pre18 it failed to receive any faxes. Ryan -- _ -- 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] sangoma card rx/tx gain level
On Apr 15, 2011, at 12:50 PM, satish patel wrote: We had echo issue before so we replaced old PRI card with Sangoma A102D HWEC. Now my question is i set rx/txgain level 0.0 default do i need to touch this value or default is best. I have read on google and people say it should around 14844 on ztmonitor for rx/tx level same. I just use milliwatt and test my default 0.0 rx/tx level and it come around 4600. Do you think i need to make it around 14844 ? I have some Sangoma cards I've been running for several years at two different locations and never had to touch the gain levels. They run beautifully at the default of 0.0. I also have some running at one location that I had to change the levels quite a bit. In that instance, I found the recommended 14844 was off by about a factor of 10. If I tried to get those levels, things were obviously way out of line, all sorts of distortion would happen. I ended up using ztmonitor, but I just watched the gauge and tuned it like audio recording equipment. I targeted about 2/3 to 3/4 up the graph as the average level. I didn't find a reliable milliwatt number to test on, so I just watched a bunch of normal calls. Once I had things working at the best level testing by ear and watching the graph, I found the value ztmonitor reported was about 4500-5000, or about 10 times lower than the info I found via google (I'm guessing I saw the same page you saw with the 14844 value). -chris www.mythtech.net -- _ -- 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] Duplicate cdr records with channel local
Hi list, I use AMI to originate calls for outbound campaigns via local channel. I use CDR records to have status of the calls such as : disposition, if call was answered by live human or answering machine, reason of call failure and make accounting based on total calls that have been made. My problem is that I have more than one CDR records for a single call. One for the Leg A and another one for Leg B. As i use cdr manager to retrieve call statistics of the campaign duplicate CDR records is really an issue. Can anybody please help me on this ? Or is there a better way to have those stats. I use asterisk trunk 1.6.2 . Your help is greatly appreciated. LIVE Every Moment, LOVE Every Day Let the nature do the rest and so NERVER GIVEUP !!! -- _ -- 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
