Re: [asterisk-users] canreinvite per route
Have canreinvite set for your internal extens. You can also have canreinvite enabled by default for all and use one or more of the 't','T','h','H','w','W' or 'L' options set in your dial commands which will override the canreinvite option and not send re-invites. cheers - Ben --- On Sat, 1/17/09, Gabriel Ortiz Lour ortiz.ad...@gmail.com wrote: From: Gabriel Ortiz Lour ortiz.ad...@gmail.com Subject: [asterisk-users] canreinvite per route To: asterisk-users@lists.digium.com Date: Saturday, January 17, 2009, 10:06 PM Can I activate/deactive the canreinvite SIP flag on the dial plan? The idea is to allow reinvite only for exten - exten calls, and not for outbound calls ___ -- 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] AGI and prepaid billing
Hi Bilal, Yes it is definitely possible. And I've done it myself for a couple of our clients. Does that answer your two questions? cheers - Ben. --- On Tue, 9/23/08, bilal ghayyad [EMAIL PROTECTED] wrote: From: bilal ghayyad [EMAIL PROTECTED] Subject: [asterisk-users] AGI and prepaid billing To: asterisk-users@lists.digium.com Date: Tuesday, September 23, 2008, 9:52 AM Hi All; Did anyone do an prepaid billing application via AGI? I would like to know if that is possible. Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] dtmf passthru
Look at the canreinvite option. - Original Message From: Rizwan Hisham [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 17, 2008 3:20:40 PM Subject: [asterisk-users] dtmf passthru hi all, Is there an option of dtmf passthru mode in asterisk. If yes, how can i do it? -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] dtmf passthru
ooopss.. I was in a hurry, wasn't i!?? what is DTMF pass thru?? As far as I know, there's nothing specific for just DTMF as pass through.. its for the entire call that is established.. for the codecs being used within the call. What is the requirement anyway? - Ben. - Original Message From: Benjamin Jacob [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 17, 2008 4:45:13 PM Subject: Re: [asterisk-users] dtmf passthru Look at the canreinvite option. - Original Message From: Rizwan Hisham [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 17, 2008 3:20:40 PM Subject: [asterisk-users] dtmf passthru hi all, Is there an option of dtmf passthru mode in asterisk. If yes, how can i do it? -- Best Regards Rizwan Hisham ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] entering a password to have access to a sip account?!
Hello Roland,
You can use the cmd Read for this.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
Pretty straight forward. Whenever you need to accept DTMF input from the user
collect the required digits using Read; check the collected digits; if yes jump
to required extension; else reject user or whatever you want to do.
I could've written out the dialplan, but well... you are a newbie you said, so
you gotta learn ;-) .
Hope this helps.
- Ben.
--- On Sun, 8/24/08, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
From: RoLaNd RoLaNd [EMAIL PROTECTED]
Subject: [asterisk-users] entering a password to have access to a sip
account?!
To: asterisk-users@lists.digium.com
Date: Sunday, August 24, 2008, 3:26 PM
Hi all,
i;m obviously a newbie, its been 2 days that im trying to
figure out a way to deny a specific extension (300) from
calling another specific extensions (03) except if the
caller punch a specified password.. sorry if im not
explaining myself well.. heres an example:
i called my pstn line(with 300 as its sip account), an
attendant answers and asks me to punch in an extension
number right now if i dial 03 it rings at the
other end! though i dont want that to happen! i want to set
asterisk up in a way tht if i dial 03 from
300 to ask me for a password... or it wont let
the line go through!
can anyone guide me through this issue! im really going
crazy to get this done! any help would truly and utterly be
appreciated:)
ps: find below my extensions.conf
[sipura-line]
exten = 301,1,Answer() ; Answer inbound calls
exten = 301,2,Playback(silence/1)
exten = 301,3,Background(simzy1) ; input an extension
exten = 301,4,WaitExten(8)
exten = 301,5,Dial(SIP/100,15) ; goes to operator
exten = 301,4,Wait(8)
include = spa
exten = _XXX,6,VoiceMail([EMAIL PROTECTED])
exten = 301,n,Hangup()
[spa]
exten =_301,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals
to 5 seconds so it will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2
voicemail box if line is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals
to 5 seconds so it will ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to
voicemail box if line is busy or unavailable
exten = _2XX,3,HangUp()
exten = _3XX,1,Dial(SIP/${EXTEN},20) ; each ring equals
to 5 seconds so it will ring 3 times
exten = _3XX,2,VoiceMail([EMAIL PROTECTED]) ; directs 2
voicemail box if line is busy or unavailable
exten = _3XX,3,HangUp()
exten =_01,1,Dial(SIP/$(EXTEN)@300) ; old ogero line
;exten =_01,2,Set(TIMEOUT(absolute)=5)
exten =_02,1,Dial(SIP/$(EXTEN)@304) ; new ogero line
exten =_03,1,Dial(SIP/$(EXTEN)@305) ; samer
exten =_04,1,Dial(SIP/$(EXTEN)@306) ; gilberte
exten =_05,1,Dial(SIP/$(EXTEN)@307) ; conference
exten =_06,1,Dial(SIP/$(EXTEN)@308) ; line 4
exten = 303,1,VoicemailMain ; voicemail box to be
redirected to
_
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Re: [asterisk-users] (no subject)
Use SendDTMF. --- On Thu, 7/3/08, Neha Punia [EMAIL PROTECTED] wrote: From: Neha Punia [EMAIL PROTECTED] Subject: [asterisk-users] (no subject) To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Date: Thursday, July 3, 2008, 10:29 AM Hi I m making a call from one asterisk server to an asterisk client The call gets completed but I want it to send dtmf signals The dialplan I have made for this is like exten = 205,1,Answer exten = 205,n,Wait(15) exten = 205,n,Playback(dtmf-1) exten = 205,n,Wait(20) but it does not send any dtmf signal where is the problem?? CAUTION - Disclaimer * This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful. This e-mail may contain viruses. Infosys has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachment. Infosys reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infosys e-mail system. ***INFOSYS End of Disclaimer INFOSYS***___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] User unable to use DTMFs?
Care to explain the scenario Vincent? Is it a SIP peer? what is the DTMF mode set? etc. --- On Tue, 7/1/08, Vincent [EMAIL PROTECTED] wrote: From: Vincent [EMAIL PROTECTED] Subject: [asterisk-users] User unable to use DTMFs? To: asterisk-users@lists.digium.com Date: Tuesday, July 1, 2008, 11:09 AM Hello A user seems unable to type DTMF in our Asterisk IVR menu. Can this be due to their phone or PBX that disables DTMFs when a user is off-hook? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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 realtime
If by realtime, you mean to be able to read the MOH class from a DB and set MusicOnHold, then I think you should try func_odbc. Have never tried it, but reading the workings of it, it seems to be possible to achieve this. Let me know if you succeed in it. - Ben. --- On Tue, 7/1/08, Nhadie [EMAIL PROTECTED] wrote: From: Nhadie [EMAIL PROTECTED] Subject: [asterisk-users] music on hold realtime To: asterisk-users@lists.digium.com Date: Tuesday, July 1, 2008, 1:33 PM Hi, Is it possible to use realtime for Music On Hold? Is it also possible to store the music/audio files on the database, same way a voicemail can be stored on the database? Thank You Regards, Nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Choppy audio
modprobe zaptel; modprobe ztdummy That will start zaptel and ztdummy after the 'zaptel stop'. Then restart asterisk. --- On Wed, 7/2/08, Doug Crompton [EMAIL PROTECTED] wrote: From: Doug Crompton [EMAIL PROTECTED] Subject: Re: [asterisk-users] Choppy audio To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, July 2, 2008, 1:58 AM OK just to be clear on what you recommend... Stop everything, unload zaptel and zrdummy modules... then just restart asterisk? Does it start zaptel? This is NOT a slow box. P6 dual core 4 gig cache, 3800 bogomips. Doug On Tue, 1 Jul 2008, bkruse wrote: I would recommend stopping asterisk (/etc/init.d/asterisk stop) /etc/init.d/zaptel stop (unload all modules) modprobe zaptel; modprobe ztdummy (in the case that you don't have another card for a timing device) /etc/init.d/asterisk start ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] start n run an agi script on hangup
I think, you can use the 'h' extension to invoke scripts (DeadAGI to be more precise) on hungup channels. use something like this : exten = _X., 1, NoOp(got a call) exten = _X., n, Dial(somexten} exten = h, 1, DeadAGI(hangupScript.sh) --- On Fri, 6/13/08, Robor Oghene [EMAIL PROTECTED] wrote: From: Robor Oghene [EMAIL PROTECTED] Subject: Re: [asterisk-users] start n run an agi script on hangup To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, June 13, 2008, 8:59 PM Thanks a million Sherwood!! On Fri, Jun 13, 2008 at 9:31 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Robor Oghene wrote: Thanks Steve, I appreciate your response, I checked the link and it talks about an agi script running before and continuing after hangup. the problem I have is that, I dont want to run an agi while the channel is up. i want to start the script on on hangup to do database cleanup.. i'd appreciate if you'd shed more light just in case am missing something.. Rgds On Fri, Jun 13, 2008 at 3:05 PM, Sherwood McGowan [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Robor Oghene wrote: hello All, How do I start and run an agi script on channel hang up? Rgds, ___ -- 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 Define the h extension in the context in question, and use DeadAGI http://www.google.com/search?hl=enq=how+do+I+run+an+AGI+after+hangupbtnG=Search http://www.google.com/search?hl=enq=how+do+I+run+an+AGI+after+hangupbtnG=Search Google is nice -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ -- 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 DeadAGI will run even after both channels are down, that's why they called it DeadAGI. VERY useful when put in the h exten :) -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- 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 ___ -- 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
[asterisk-users] mute a call/ re-invite mid-session?
Hello ppl, Is there anyway to control a call mid-way in terms of sending a re-INVITE with say sendonly, etc. to mute one call leg of a bridged call ?? Looked around, so far, doesnt seem to be possible. If it's not, I think it's quite an important feature (re-INVITES mid-session) for a B2BUA. cheers - Ben. ___ -- 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] update DB on ringing/ catch ringing event
Update on this one.
I finally went back to AMI only for implementing this particular feature, but
ofcourse I had to make an addition of a couple of lines for my particular
requirement.
On Dial, the 'dial' event is sent over AMI which I capture. Unfortunately the
event didn't have any field identifying the account/or other user settable data
for that particular call. So, I added lines in app_dial.c to send even the CDR
userfield in the event.
So, before doing the 'Dial' I set CDR userfield with my own data, which is
captured by the AMI user and populates/updates the correct row in my DB with
the dialed channel, etc.
From this point on, I can hangup the required channel, even before it has been
answered/ even before it has started ringing.
static void senddialevent(struct ast_channel *src, struct ast_channel *dst)
{
manager_event(EVENT_FLAG_CALL, Dial,
Source: %s\r\n
Destination: %s\r\n
CallerID: %s\r\n
CallerIDName: %s\r\n
SrcUniqueID: %s\r\n
DestUniqueID: %s\r\n
CDRUserfield: %s\r\n,
src-name, dst-name, src-cid.cid_num ? src-cid.cid_num : unknown,
src-cid.cid_name ? src-cid.cid_name : unknown, src-uniqueid,
dst-uniqueid,
(dst-cdr)?(dst-cdr-userfield):);
}
I am writing this mail from home, so don't really have the exact field names.
cheers
- Ben.
--- On Thu, 5/8/08, Tzafrir Cohen [EMAIL PROTECTED] wrote:
From: Tzafrir Cohen [EMAIL PROTECTED]
Subject: Re: [asterisk-users] update DB on ringing/ catch ringing event
To: asterisk-users@lists.digium.com
Date: Thursday, May 8, 2008, 12:00 AM
On Thu, May 08, 2008 at 12:19:52AM +0300, Atis Lezdins
wrote:
On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Benjamin Jacob schrieb:
Anyway in Asterisk to update a DB/ do some
action on
events like ringing.
The issue is I need to be able to
hangup/cancel a
call, if it's ringing(decided by the
admin). This is
independant of the timeout that we can
specify in the
Dial command.
If I could somehow update a DB with the
channel name
on ringing, it would solve my problem.
I assume NVlinedetect is one way to do it,
but that
isn't visible anymore, more so for
Asterisk 1.4 and
above.
Any bright ideas on this one?
I think there is no other solution but to listen
to events on
the Asterisk manager interface.
For now, not really.
You could try Realtime Channels patch I just mentioned
here:
http://lists.digium.com/pipermail/asterisk-users/2008-May/211136.html
This should give you up-to-date list of channels in
database, so you can use
SELECT * FROM channels WHERE state=Ring;
to get currently ringing channels.
If You find this patch useful, please add a comment to
issue
http://bugs.digium.com/view.php?id=12556
that you would like to see Realtime status implemented
in future
versions of Asterisk.
So you constantly poll the status of all channels? Waiting
on manager
interface event sounds more effective to me.
But what exact ringing is it? Isn't the
call by then already in the
dialplan (and could be hung up before answered?)
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] are channel names unique
Hello ppl, Are the channel names generated on 'Dial's supposed to be unique? I see the channel names repeating on my asterisk box. I just wanted to confirm this. Can anyone point me to the lines of code where the channel name is generated/calculated? I tried looking, but it looks like quite a big maze. Regards - Ben. ___ -- 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] are channel names unique
So I thought!! Thanks guys.
But a query with regards to this :
I need to send hangup commands based on these channel names only. So at any
given point of time, for 'n' ongoing calls, will these 'n' channel names be
different/ unique?
If not, using AMI, how do we hangup a given channel?
cheers
- Ben.
--- On Thu, 5/15/08, Russell Bryant [EMAIL PROTECTED] wrote:
From: Russell Bryant [EMAIL PROTECTED]
Subject: Re: [asterisk-users] are channel names unique
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Thursday, May 15, 2008, 11:46 AM
Benjamin Jacob wrote:
Are the channel names generated on 'Dial's
supposed to be unique?
I see the channel names repeating on my asterisk box.
I just wanted to confirm this.
Can anyone point me to the lines of code where the
channel name is generated/calculated? I tried looking, but
it looks like quite a big maze.
Channel names are not guaranteed to be unique at all.
However, all channels
have a uniqueid associated with them. You can access it in
the dialplan via
${UNIQUEID}.
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
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[asterisk-users] update DB on ringing/ catch ringing event
Hello ppl, Anyway in Asterisk to update a DB/ do some action on events like ringing. The issue is I need to be able to hangup/cancel a call, if it's ringing(decided by the admin). This is independant of the timeout that we can specify in the Dial command. If I could somehow update a DB with the channel name on ringing, it would solve my problem. I assume NVlinedetect is one way to do it, but that isn't visible anymore, more so for Asterisk 1.4 and above. Any bright ideas on this one? cheers - Ben. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] simple realtime question
Last I was working on it, it did indeed NOT look at sip.conf with realtime architecture being used. But why take chances anyway? Move all the relevant conf files from /etc/asterisk to some other place to be safe. cheers - Ben. --- Rilawich Ango [EMAIL PROTECTED] wrote: HI, Does asterisk will ignore the setting in files if realtime is applied, say asterisk will ignore all the setting in sip.conf if realtime table sip_buddies is applied? ango ___ -- 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 Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] Asterisk - get Caller String(as per key action)
Hiren, Not really clear as to what are the things you exactly want. List them out clearly. Before you do that, do google and read up on Asterisk+IVR Asterisk+agi Your need to calculate sum of birthdate digits etc can be achieved using AGI scripting. cheers - Ben. --- Hiren Mistry [EMAIL PROTECTED] wrote: Dear Sir, I have a one query for Asterisk, I want to make a dial plan to the conference in Caller, Asterisk and my staff, and my staff will also transfer call to return PBX to IVR. and when caller make press key as a date of birth then ivr can make calculate sum in one digite (Example :- 12/12/2000 total is 8 ). So what I have to do in my dial-plan. My Asterisk System is transfer call to PSTN line. With Regards, Hiren Mistry. ___ -- 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 Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy
http://www.openvox.com.cn/products_detail.php?genre_id=9id=28 If you can get the bare card, you can use it for timing with a little magic that can be found via google. If not, get one with an FXO or FXS and you will add a little flexibility and have real hardware timing. If you continue to have issues, then you can eliminate timing and focus on processes I would think. I had a client running spamassassin on their Asterisk box which doubled as their corporate email server, geewhiz, I wonder why they were having issues. Another odd thing Tzafrir helped me to notice was (I don't remember what version of CentOS) that the time was jumping ahead a couple of minutes and then back. Running top, you could tell something was up because it was refreshing way too fast. Then typing date on the command line repeatedly showed the time jumping all over the place. Might want to check that out too. Thanks, Steve Totaro Thanks again guys. the 'watch -d -n 1 cat /proc/interrupts' showed things to be ok.. the rtc cycles increasing by 1024+ per second. In the process of cleaning up unnecesary processes, I came across this line : /usr/sbin/vmware-guestd --background /var/run/vmware-guestd.pid GASP so does this mean this is a virtual machine?? I have got no idea about virtualization yet. So how do I confirm if this is a virtual machine or not?? And is it advised to run asterisk on a virtual machine? - Ben. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy
- In the process of cleaning up unnecesary processes, I came across this line : /usr/sbin/vmware-guestd --background /var/run/vmware-guestd.pid GASP so does this mean this is a virtual machine?? I have got no idea about virtualization yet. So how do I confirm if this is a virtual machine or not?? And is it advised to run asterisk on a virtual machine? - Ben. Ben, Who is providing your server? I assume it is in a colo. Ask them or see if they mention it in their agreement or sales material. Finally, you can just ask them. If they claimed a dedicated server, complain. It seems you are running on a virtual machine and no, that is not advisable. Thanks, Steve Totaro Steve, Naa.. it's not co-lo. It's a dedicated server for sure, but my client wants to make the most out of one box, it seems. Talked to the client today and confirmed that it is indeed a virtual machine. They said they had previously installed asterisk around a year back on a virtual machine with no issues. I did not have any solid convincing response to that. I do understand about virtualization not being a recommended thing to do. Now to convince the client. Also, if I put in a fxo/fxs card, i've read somewhere that virtual machines won't be able to access the card n hence the timing provided by it. Is it true? thanks again Steve and the rest of you. - Ben. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy
OK, I think you need to home in on the differences between the server(s) that work fine and the one that doesn't. As I said in my other mail, the faulty one is a .. mono processor machine, with SMP turned on .. running CentOS 5 .. with kernel : 2.6.18-53.1.13.el5 There are other kernels too(2.6.18-8, etc.), will be trying those kernels too. The local working machine is : .. dual processor, with SMP ofcourse .. running Fedora Core 7, if I remember it correctly. .. kernel definitely 2.6.13 Have looked at all parameters, be it the kernel timer frequency(1000 HZ), enhanced timer support, etc. Everything seems to be set right. (Then again, I hope I am looking at the correct places, i.e. .config files and using make menuconfig). Try watch -dn 1 cat /proc/interrupts and check that the RTC interrupts are going up by 1024 per second. This is with ztdummy running. This I gotta try. What if it isn't? And worse, what if it is and I am still getting the choppy playbacks!! What else is going on on this server? Does it have any virtual machines on it? Does it have X Windows running? What does top show? Unfortunately a lot of other processes are running too on the server, one of them being httpd and other sundry needed by the client (this inspite of suggesting him to otherwise). This is an Asterisk install not done by me, I just added the zaptel installation and ztdummy module. Was brazenly confident of things working in a jiffy(does this count as a pun?), when I stepped in. cheerz :-( - Ben. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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
[asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy
Hello ppl, One on my clients' machine had Asterisk 1.4.4. installed. The complained of choppy Playback of gsm files. So scouring the internet gave me the solution of installing ztdummy and loading it as a module. Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and re-installed. Sill no effect. Do I have to specify any parameter in the Asterisk compilation to look at ztdummy/rtc? As far as I remember (am coming back to Asterisk after quite some time now), you don't really need to set anything over there for any zaptel specific compilation? And yes, all the files are gsm files and the codec used for the calls is ulaw. I even tried converting those gsm files to wav using sox and then playing them, but the behaviour is the same. Any ideas anyone.. something I am missing ?? TiA, - Ben. - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- 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
[asterisk-users] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy
Are my messages getting through? This is urgent!! Any pointers? Benjamin Jacob [EMAIL PROTECTED] wrote: Date: Thu, 24 Apr 2008 23:23:08 -0700 (PDT) From: Benjamin Jacob [EMAIL PROTECTED] Subject: Playback / Background / Read choppy, but musiconhold fine, even with ztdummy To: asterisk-users@lists.digium.com Hello ppl, One on my clients' machine had Asterisk 1.4.4. installed. The complained of choppy Playback of gsm files. So scouring the internet gave me the solution of installing ztdummy and loading it as a module. Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and re-installed. Sill no effect. Do I have to specify any parameter in the Asterisk compilation to look at ztdummy/rtc? As far as I remember (am coming back to Asterisk after quite some time now), you don't really need to set anything over there for any zaptel specific compilation? And yes, all the files are gsm files and the codec used for the calls is ulaw. I even tried converting those gsm files to wav using sox and then playing them, but the behaviour is the same. Any ideas anyone.. something I am missing ?? TiA, - Ben. - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- 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] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy
Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Benjamin Jacob wrote: One on my clients' machine had Asterisk 1.4.4. installed. The complained of choppy Playback of gsm files. So scouring the internet gave me the solution of installing ztdummy and loading it as a module. Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and re-installed. Sill no effect. Do I have to specify any parameter in the Asterisk compilation to look at ztdummy/rtc? As far as I remember (am coming back to Asterisk after quite some time now), you don't really need to set anything over there for any zaptel specific compilation? And yes, all the files are gsm files and the codec used for the calls is ulaw. I even tried converting those gsm files to wav using sox and then playing them, but the behaviour is the same. Any ideas anyone.. something I am missing ?? Firstly, check whether Asterisk has chan_zap loaded and access to zaptel: *CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefault *CLI If you don't get pseudo shown, then you are not getting the benefit of ztdummy. However, the probably main cause of choppy sound is poor timing from the SIP client (I'm assuming SIP), because Asterisk by default uses the incoming stream to generate timing for the outbound stream. There are two main things to try: 1. Make sure that the SIP clients are NOT using silence suppression (may be referred to as VAD, bandwidth saving, or something similar). 2. If ztdummy is running ok, edit /etc/asterisk/asterisk.conf and enable the line internal_timing=yes. That should make it play out based on internal zaptel timing instead of timing off the incoming stream, I think. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org Thanks Tony for the response. zap show channels shows that things are fine, as you said : *CLI zap show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault Tried setting internal_timing to yes as well. Still no difference. Also, I don't think my SIP gateway uses Silence suppression, because the same SIP gateway connections work fine with another Asterisk server. This is getting seriously irritating now!!! Have tried all the tricks and tips I've been finding on the net. Yeah, btw, even Meetme playback is choppy. So, I think its somehow related to timing. But I am not the expert. - Ben. - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- 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] Playback / Background / Read choppy, but musiconhold fine, even with ztdummy
Benjamin Jacob [EMAIL PROTECTED] wrote: Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Benjamin Jacob wrote: One on my clients' machine had Asterisk 1.4.4. installed. The complained of choppy Playback of gsm files. So scouring the internet gave me the solution of installing ztdummy and loading it as a module. Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and re-installed. Sill no effect. Do I have to specify any parameter in the Asterisk compilation to look at ztdummy/rtc? As far as I remember (am coming back to Asterisk after quite some time now), you don't really need to set anything over there for any zaptel specific compilation? And yes, all the files are gsm files and the codec used for the calls is ulaw. I even tried converting those gsm files to wav using sox and then playing them, but the behaviour is the same. Any ideas anyone.. something I am missing ?? Firstly, check whether Asterisk has chan_zap loaded and access to zaptel: *CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefault *CLI If you don't get pseudo shown, then you are not getting the benefit of ztdummy. However, the probably main cause of choppy sound is poor timing from the SIP client (I'm assuming SIP), because Asterisk by default uses the incoming stream to generate timing for the outbound stream. There are two main things to try: 1. Make sure that the SIP clients are NOT using silence suppression (may be referred to as VAD, bandwidth saving, or something similar). 2. If ztdummy is running ok, edit /etc/asterisk/asterisk.conf and enable the line internal_timing=yes. That should make it play out based on internal zaptel timing instead of timing off the incoming stream, I think. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org Thanks Tony for the response. zap show channels shows that things are fine, as you said : *CLI zap show channels Chan Extension Context Language MOH Interpret pseudodefaultdefault Tried setting internal_timing to yes as well. Still no difference. Also, I don't think my SIP gateway uses Silence suppression, because the same SIP gateway connections work fine with another Asterisk server. This is getting seriously irritating now!!! Have tried all the tricks and tips I've been finding on the net. Yeah, btw, even Meetme playback is choppy. So, I think its somehow related to timing. But I am not the expert. - Ben. - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- 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-usersBtw, I am on CentOS 5, with uname showing as: Linux mserver.org 2.6.18-53.1.13.el5 #1 SMP Tue Feb 12 13:01:45 EST 2008 i686 i686 i386 GNU/Linux And it is not a multiprocessor machine. Will the SMP option affect the working in any way? - Ben. - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- 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] re-invite (bypass asterisk) post call establishment
Hi again, I tried this again, but the reInvite happens immediately after the 200 OK/ACK. And then the D() specified DTMF is sent. Attached is the SIP trace for the calls. I call (from Asterisk) - 0119198807x After connect, I dial - 31927x. This number 31927x is the conference bridge and I need to send DTMF (the bridge PIN) to it after connection. But alas, the reinvite happens before the D() is executed. The SIP gateway is MySIPGateway at 204.aaa.bbb.ccc. cheers - Ben. Steve Davies [EMAIL PROTECTED] wrote: 2008/4/22 Benjamin Jacob : [snip] So, my question : once the SDPs are exchanged, what will happen to the DTMFs sent by Asterisk using sendDTMF or the D option in dial. [snip] As far as I can tell, the D() option will be executed before the re-invite takes place, so Asterisk will still be in-line. I believe that the dial is not considered complete/connected until the D() is finished. Cheers, Steve ___ -- 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 - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. reInvite Description: 1957794313-reInvite ___ -- 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
[asterisk-users] sip channel - detect ringing (nvlinedetect??)
Hello ppl, Is there any other way to detect states like Ringing on SIP channels on Asterisk? Nvlinedetect is one way, but it seems to have disappeared from the face of the earth! Any pointers or does anyone have the code for NV* features? Thanks in advance - Ben. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ___ -- 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
[asterisk-users] API Originate - action on reject/busy/congestion
Hello ppl, I am using the Astman API Originate command to initiate a call to a user. On connect of the user, I dial another user to bridge the call between the two. I am using the Async option with the Originate command, as I don't want to use Astman proxy yet. Is there any way to invoke a script, etc if the first user doesn't pick up the call/rejects it or we get a congestion on that channel? TiA, - Ben. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ___ -- 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
[asterisk-users] re-Invite post call establishment (for RTP bypass)
Hello ppl, Any way to do a re-invite and make RTP bypass Asterisk, after call establishment. In other words, I would like to control when to do the bypass work for peer-peer RTP flow. The issue is that I need to send DTMFs after dialing the user because most of the users are behind PBXes (having individual extensions) themselves and almost all of the PBXes send a 200 OK and then play out the PBX messages. So I need to send the extension DTMFs first, bridge the calls and then re-invite users for them to do a peer-peer rtp conversation. TiA, - Ben. - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- 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
[asterisk-users] re-invite (bypass asterisk) post call establishment
Hello ppl, Any way to do a re-invite and make RTP bypass Asterisk, after call establishment. In other words, I would like to control when to do the bypass work for peer-peer RTP flow. The issue is that I need to send DTMFs after dialing the user because most of the users are behind PBXes (having individual extensions) themselves and almost all of the PBXes send a 200 OK and then play out the PBX messages. So I need to send the extension DTMFs first, bridge the calls and then re-invite users for them to do a peer-peer rtp conversation. TiA, - Ben. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ___ -- 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] re-invite (bypass asterisk) post call establishment
Apologies for not explaining the set up . Using AstMan API, I Originate a call to user A. User A is a conference bridge which needs pin authentication. So post 200 OK, I need to send DTMFs for that pin. After sending the pin, I Dial (using the Originate context) user B. Now user B is behind a PBX, so I need to dial the extension for user B. I send the extension digits using DTMFs again. So, if I set canreinvite=yes, as soon as I get a 183/200 OK from user B, re-Invites are sent to both participants with the other's SDP. So, my question : once the SDPs are exchanged, what will happen to the DTMFs sent by Asterisk using sendDTMF or the D option in dial. Another scenario would be to call user B first and then user A first. The same case applies over there as well. Is there any other way to tell asterisk when to do a re-Invite/control the timing of the re-Invite? Hope I am clear this time. cheerz - Ben. Steve Davies [EMAIL PROTECTED] wrote: On 21/04/2008, Benjamin Jacob wrote: Hello ppl, Any way to do a re-invite and make RTP bypass Asterisk, after call establishment. In other words, I would like to control when to do the bypass work for peer-peer RTP flow. The issue is that I need to send DTMFs after dialing the user because most of the users are behind PBXes (having individual extensions) themselves and almost all of the PBXes send a 200 OK and then play out the PBX messages. So I need to send the extension DTMFs first, bridge the calls and then re-invite users for them to do a peer-peer rtp conversation. TiA, - Ben. You don't say what you've tried already, but as long as canreinvite=yes is set against the SIP peer, the RTP stream should be redirected once the connection is open. As far as DTMF to dial an extension at the remote end, have you looked at the D() parameter to the Dial command? Regards, Steve ___ -- 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 - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- 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] Realtime: Should I say or should I go (now) ?
I am using Realtime in virtually all my projects. So far, I haven't had any major issues. It saves a lot of headache for profile/dialplan updates, at least for me! So I say, GO! - Ben Olivier wrote: Hi, I'm working on a 500 seats Asterisk project. I'm wondering whether or not I should consider using Asterisk Realtime and a database to manage phones registrations. Stories in Dev mailing list say Realtime is mis-used or should be improved. So, what's the bottom line ? Can I consider anything I can do with .conf files can be done with a combination of .conf files and Realtime. Regards ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] VoIP service providers/PSTN termination points
Thanks good ppl! Doug wrote: At 10:02 12/17/2007, mail-lists wrote: Same here - Gafachi has been great. Decent rates, very stable and great voice quality. I use Gafachi.com http://Gafachi.com and have good quality with no minimum requirements. Try them at www.gafachi.com http://www.gafachi.com Triple Ditto for Gafachi. ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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
[asterisk-users] VoIP service providers/PSTN termination points
Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere http://www.iconnecthere.com Vonage http://www.vonage.com Teliax http://www.teliax.com I found something known as Inphonex http://www.inphonex.com. These had the cheapest rates and quite a good coverage too. Anyone with experience on this one? I am looking at a combination of decent prices and good quality. Any other suggestions or ideas welcome too. TiA - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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
[asterisk-users] [Fwd: load test zap channels (in and out)]
Is this getting through?? EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ---BeginMessage--- Hello ppl, Am totaly new to this zap thingy.. zapped I would say I am! (couldn't resist that cliche...). Just like sipp for testing SIP channels, do we have any such tools to test zap channels? I did investigate a bit and understood a bit. To generate enough number of calls, I can use the call files or something similar. . Have a T1 card on the box, so what are the elements required outside the box, i.e. for the termination of calls originated from the Asterisk box. . What are the limits as to the number of simultaneous calls per channel? . Any tools/boxes to generate 'n' number of calls(/sec and so on) to this T1 interface on * ? Also, when reading about zap, got lines like : Dial(Zap/1-2/c1234) ; Dial 1234 on span 1, port 2, with PRI clear indication. What does span 1, port 2 mean? Thanks in advance - Ben. ---End Message--- ___ --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] retrieve last number dialled
simultaneous calls??.. will this correctly ensure the last call retrieved from such DB was indeed the last call received? Patrick wrote: On Wed, 2007-11-28 at 11:07 +0100, Eric Smith wrote: What is the easiest (simplest) way to do this? Store the dialed number in the Asterisk DB and setup an extension to retrieve it from the DB and dial it. Regards, Patrick ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] retrieve last number dialled
duhhh !! Patrick wrote: On Wed, 2007-11-28 at 17:08 +0530, Benjamin Jacob wrote: simultaneous calls??.. will this correctly ensure the last call retrieved from such DB was indeed the last call received? Look at the subject. He said *dialled* number, not received :) Regards, Patrick ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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
[asterisk-users] Billing/Call Control engine : AGI scripts/ AstMan API
Hello ppl, Have implemented a really nice Billing engine using AGI scripts. So far it works fine, tho haven't yet put it in the torture cell. The AGI scripts have been written in PHP, using MySQL for the billing and profile information. The major disadvantages I see using AGI scripts : 1. A new process(invocation of PHP scripts) on every new call. 2. MySQL connections on every instance of the PHP AGI script. (I am not too sure, if connections can be maintained across processes, am no PHP guru. I think, if I write in C/C++ can use shared memory for maintaining the connection). So, to overcome these issues, I was thinking of using AstMan APIs along with astmanproxy, with the setup being something like this : Asterisk - astmanproxy - Billing Engine(control/access) Has anyone ever tried this? The one seriously big work with this approach would be to have an FSM built into my billing engine, maintaining call states, etc. That seems to be quite a daunting task to be done in a short time. Any ideas anyone?or any similar experiences, in terms of performance, scalability, etc. w.r.t both AGI scripts and AstMan API? TiA - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] Phones Not Registering
The reason could be bad routing, IPs used by multiple devices.. n so on... Edwin Kariuki wrote: Hi, I have a voip platform that has a SIP server where about 450 sipura phones adaptors register. On two occassions some phones (which were previously working) have refused to register with certain IPs but when I change the IP the phones register. The failing IP can the work after two days. A trace from the server shows that the phone is sending a registration signal to the server that the server is also sending back the same but its not getting to the phone. What could be the cause of this? Thanks, Edwin Get easy, one-click access to your favorites. Make Yahoo! your homepage. http://us.rd.yahoo.com/evt=51443/*http://www.yahoo.com/r/hs ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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
[asterisk-users] common/shared voicemail box
Hello All, I am using ODBC storage for voicemail on my asterisk box. I want to have a common voicemail box for different extensions. I know how to do that, but the question troubling me is how and where do I store the the extension name for which a particular voicemail was left. e.g. extensions 1000, 1001, 1002 all using the same voicemailbox 5. Now, when someone calls 1000, and leaves a voicemail, I want to store the fact that this voicemail was meant for extension 1000. Similarly for 1001 and so on. Any ideas anyone? TiA - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] DTMF Problem
for UDP tcpdump -nnXs 0 udp -i eth0 -w name.cap Btw, a pcap file (created on a linux server using tcpdump) capturing the RTP(udp) traffic opened up in wireshark, wireshark doesn't really format(or recognize) the packets as RTP, unlike the capture done live from a wireshark configured to capture RTP traffic. In the former, wireshark shows up everything as UDP and I have to do a lot of manual parsing to find out the type etc in the packets captured. Am I missing some config on wireshark here? TiA - Ben. ľľ wrote: You can use the tcpdump comand in linux. Like: tcpdump -i eth0 -s 0 -w name.cap And you can open the cap file useing wireshark that is a good 木木 2007-11-16 *发件人：* Doug *发送时间：* 2007-11-16 00:53:15 *收件人：* Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - No *抄送：* *主题：* Re: [asterisk-users] DTMF Problem At 06:42 11/15/2007, =?gb2312?B?xL7Evg==?= wrote: Hi， Could you capture the the UDP package How is this done? in all of your server, Asterisk A, Asterisk B, ser, Asterisk C. And you can find that server who lost the DTMF (RTP EVENT). -- Amy 2007-11-15 -- 发件人： Arun Kumar 发送时间： 2007-11-15 20:30:45 收件人： Asterisk Users Mailing List - Non-Commercial Discussion; SER Users 抄送： 主题： [asterisk-users] DTMF Problem Hi Here is my setup: USER -- PSTN - Asterisk A IAX2 Trunk Asterisk B - SER Asterisk C I'm not able to receive DTMF passed by USER on Asterisk C. All my asterisk boxs are configured with same DTMF type (auto) but no luck. Please help on this issue. Thanks, Arun ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] Problem with AGI Script
Steve Edwards wrote: On Thu, 15 Nov 2007, Benjamin Jacob wrote: well.. if nothings working.. try putting in debug lines urself in the code.. say use system calls to write some debugging data into some temporary file in ur perl code. I'm a big fan of syslog(LOG_ERR, I expected %d, but I got %d, foo, bar); to write a message to the system log. A single statement and no temporary files to clean up. Syslog has lots of features -- check out the man page. thats definitely better.. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] Problem with AGI Script
well.. if nothings working.. try putting in debug lines urself in the
code.. say
use system calls to write some debugging data into some temporary file
in ur perl code.
let us know..
Matt wrote:
[EMAIL PROTECTED] agi-bin]# /usr/bin/perl -v
This is perl, v5.8.5 built for i386-linux-thread-multi
Debug shows nothing:
-- Launched AGI Script /var/lib/asterisk/agi-bin/GetEmailfromDID.agi
AGI Tx agi_request: GetEmailfromDID.agi
AGI Tx agi_channel: Zap/23-1
AGI Tx agi_language: en
AGI Tx agi_type: Zap
AGI Tx agi_uniqueid: 1195061174.4
AGI Tx agi_callerid: 5706016716
AGI Tx agi_calleridname: Test Networks
AGI Tx agi_callingpres: 0
AGI Tx agi_callingani2: 0
AGI Tx agi_callington: 33
AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: 5706010280
AGI Tx agi_rdnis: unknown
AGI Tx agi_context: macro-faxreceive
AGI Tx agi_extension: s
AGI Tx agi_priority: 2
AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode:
AGI Tx
-- AGI Script GetEmailfromDID.agi completed, returning 0
Just returned with a 0 and doesn't do anything it is suppose to do.
I'm kind of at a loss.
On Nov 14, 2007 11:40 AM, Mindaugas Kezys [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Make sure /usr/bin/perl can be reached.
Also try in your CLI:
agi debug
Same case happens when I do not have php-cli installed for php AGI
scripts.
Mindaugas Kezys
http://www.kolmisoft.com
MOR – Advanced Billing for Asterisk PBX
*From:* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]] *On Behalf Of *Matt
*Sent:* Wednesday, November 14, 2007 4:00 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Problem with AGI Script
I have asterisk 1.2.18 running on a new system we just installed.
Although I've used AGIs many times in the past, I'm stumped on
this one. It may just be a simple issue that I need another eyeset
to look at.
My AGI does the following:
#!/usr/bin/perl
#Load a few modules...
use Asterisk::AGI;
use DBI;
$AGI = new Asterisk::AGI;
#Grab input from Asterisk
my %input = $AGI-ReadParse();
#Some Debugging
$AGI-exec('SayDigits',$ARGV[0]);
exit;
All seems fine. If I run the script from the command line it works
as expected:
[EMAIL PROTECTED] agi-bin]# ./GetEmailFromDID.agi 333
EXEC SayDigits 333
However, when actually running in practice I get:
-- Executing AGI(Zap/23-1, GetEmailfromDID.agi|5706016716) in
new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/GetEmailfromDID.agi
-- AGI Script GetEmailfromDID.agi completed, returning 0
extensions.conf
[macro-faxreceive]
exten = s,1,Set(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,agi(GetEmailfromDID.agi|${CALLERID (number)})
exten = s,3,rxfax(${FAXFILE})
exten = s,104,Set([EMAIL PROTECTED]
mailto:[EMAIL PROTECTED])
exten = s,105,Goto(3)
Any thoughts on why asterisk doesn't seem to be passing anything
to the script and the script doesn't seem to be passing anything
back? When I call I do not hear the digits read to me, instead I
just get thrown to the next object after the digit reading.
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Re: [asterisk-users] get egress SIP call Id
Hello Steve,
I think Ray was talking more like the following setup (do correct me if
I am wrong):
User A (SIPcallId1) --- Asterisk (SIPcallId2) -- User B
In this case, the INVITE SIP callId received by Asterisk from User A is
different to that sent in the INVITE to User B.
I can get User A's callId using ${SIPCALLID}. How about accessing SIP
callid of the INVITE sent to User B??
Typical need for this, is to store both the callIds to store in the CDRs
for debugging purposes(w.r.t. the service provider, et al).
cheerz
- Ben.
Steve Totaro wrote:
You can capture the sipcallid from the manager output. The cool part is
that the sipcallid is the same on both sides of a call. So,
AsteriskA---SIP (sipcallid) AsteriskB SIP (Same sipcallid as
AsteriskA for that call.
It is really easy to capture it from the manager.
Thanks,
Steve
Ray Chen wrote:
Hi Philipp,
Thank you for your response to my question. I am working on a
project which uses Asterisk as the voice engine. I need to
get the ingress and egress sip call id for a call to write call CDR.
(Asterisk CDR does not meet our customer requirments). If there is
no any easy way to get it I might need to create a seperate
process/thread to query manager interface as you mentioned. Thanks you,
Ray
Ray Chen wrote:
Hi, Does anybody know how to get the SIP call ID of a Dial
command?
There's no easy way to do it. What's your
intention? There are several events on the
manager interface.
Regards,
Philipp Kempgen
--
T
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by reply email and then destroy the message. Opinions, conclusions and other
information in this message that do not relate to official business of Mascon
shall be understood to be neither given nor endorsed by Mascon. Any information
contained in this email, when addressed to Mascon clients is subject to the
terms and conditions in governing client contract.
Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we
can not guarantee that any email or attachment is free from computer viruses
and you are strongly advised to undertake your own anti-virus precautions.
Mascon grants no warranties regarding performance, use or quality of any e-mail
or attachment and undertakes no liability for loss or damage, howsoever caused.
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Re: [asterisk-users] get egress SIP call Id
Also, how do you acces the second SIP call ID from the dialplan? Any
simple way to do this?
Benjamin Jacob wrote:
Hello Steve,
I think Ray was talking more like the following setup (do correct me
if I am wrong):
User A (SIPcallId1) --- Asterisk (SIPcallId2) -- User B
In this case, the INVITE SIP callId received by Asterisk from User A
is different to that sent in the INVITE to User B.
I can get User A's callId using ${SIPCALLID}. How about accessing SIP
callid of the INVITE sent to User B??
Typical need for this, is to store both the callIds to store in the
CDRs for debugging purposes(w.r.t. the service provider, et al).
cheerz
- Ben.
Steve Totaro wrote:
You can capture the sipcallid from the manager output. The cool part
is that the sipcallid is the same on both sides of a call. So,
AsteriskA---SIP (sipcallid) AsteriskB SIP (Same sipcallid as
AsteriskA for that call.
It is really easy to capture it from the manager.
Thanks,
Steve
Ray Chen wrote:
Hi Philipp,
Thank you for your response to my question. I am working on a
project which uses Asterisk as the voice engine. I need to
get the ingress and egress sip call id for a call to write call CDR.
(Asterisk CDR does not meet our customer requirments). If there is
no any easy way to get it I might need to create a seperate
process/thread to query manager interface as you mentioned.
Thanks you,
Ray
Ray Chen wrote:
Hi, Does anybody know how to get the SIP call ID of a Dial
command?
There's no easy way to do it. What's your
intention? There are several events on the
manager interface.
Regards,
Philipp Kempgen
--
T
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whom they are addressed. Any unauthorised distribution or copying is strictly
prohibited. If you receive this transmission in error, please notify the sender
by reply email and then destroy the message. Opinions, conclusions and other
information in this message that do not relate to official business of Mascon
shall be understood to be neither given nor endorsed by Mascon. Any information
contained in this email, when addressed to Mascon clients is subject to the
terms and conditions in governing client contract.
Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we
can not guarantee that any email or attachment is free from computer viruses
and you are strongly advised to undertake your own anti-virus precautions.
Mascon grants no warranties regarding performance, use or quality of any e-mail
or attachment and undertakes no liability for loss or damage, howsoever caused.
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[asterisk-users] maximum retries exceeded on transmission Warnings
Hello All, I've got the following warning messages a couple of days back: /chan_sip.c: Maximum retries exceeded on transmission SIPcallId for seqno 1 (Critical Response). /Have got the warnings repeatedly for one Callid. If maximum retries have exceeded why should it give me those warnings again n again for the same callid, with a gap 4 seconds between each warning. The callids mentioned in the warnings are of the inbound leg. I've scoured the net, but haven't got anything conclusive. Have found responses ranging from firewall issues, no reception of ACKs, to bugs in some versions of Asterisk. I am using Asterisk 1.4.4, all SIP calls, with PSTN termination provided by my service provider. Have no firewalls or iptables set on my server. The calls did not seem to work even across a restart of asterisk. Interestingly, the calls to and from the very same numbers worked later on the next day. Anyone faced similar problems and was able to get the root of it? Or is it a bug? cheerz - Ben EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] running twice
show us the output of ur top command Pezhman Lali wrote: Dear I am using an asterisk 1.2.7.1 , with postgres and safe_Asterisk, for running, asterisk. but there is a problem, after 2-3 hours after restarting any things, top shows me, that, two asterisk, are now running, and one of them, gets 99.7 percent of cpu. Do you have any idea? Best Mani Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] prepaid application recommendation
a2billing so far seems to be quite comprehensive compared to the other freeware asterisk-based billing solutions available out there. We are building our own billing solution(due to the very peculiar requirements, one of which is to bill the callee, rather than the caller). We are achieving this so far, using AGI scripts, tho we plan to migrate to Asterisk Manager APIs soon. I haven't been able to go thru a2billing in detail(just skimmed thru the code n sample dialplans). It seems , if I am not mistaken, in a2billing every call invokes an AGI script. So this sounds a lil inefficient, where db connections, data structures etc, are created every time. Any experiences on the performance of both a2billing and AstMan API based solutions? cheerz - Ben. Sarfaraz Chougule wrote: I would recomend using Areski's billing solution : http://www.areski.net/a2billing On 9/21/07, *Rilawich Ango* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I am looking for a prepaid application. I found that there are many applications in the page http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications. Anyone recommendation among them? ango ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- With Best Regards, ** Sarfaraz Chougule ** ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Linux limits
safe_asetrisk bundled with the package, does increase the file limits in quite a neat way, with some other good setups. Edit MAXFILES or SYSMAXFILES as required. Also, I've read posts online, advising not to use safe_asterisk. Any experiences on this one, anyone? cheers - Ben. Jay R. Ashworth wrote: On Tue, Sep 18, 2007 at 04:22:29PM -0400, Alex Balashov wrote: On Tue, 18 Sep 2007, Wai Wu wrote: Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for asterisk1/700 Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in Linux itself. [ top posting fixed so I can comment as well ] You have to increase the amount of available file descriptors per process: http://hausheer.osola.com/docs/11%C2%A0%C2%A0 These days, I beleve the typical place to fix that is actually in /etc/sysctl.conf, in most distros: http://www.cs.wisc.edu/condor/condorg/linux_scalability.html That page notes it for RedHat derived distros, but I'm pretty sure SuSe puts it there as well. Cheers, -- jra EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] alphabetical extension patterns
You The Man, Anselm. Thanks for the details.
Anselm Martin Hoffmeister wrote:
Am Dienstag, den 11.09.2007, 17:11 +0530 schrieb Benjamin Jacob:
Thanks Anselm. This does clears a few things for me.
Tho, I couldnt find the patterns you mentioned in the docs(do point me
to the location if you know of it).
I started on
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
Patterns have to begin with _, meaning it is a pattern. A . stands
for one or more characters, so I only allow three-and-more character
SIP phone numbers like [EMAIL PROTECTED], but not [EMAIL PROTECTED] This
is deliberate: I rather not have catchall-type phone numbers, I already
get enough mail spam on the few catchall-addresses I have (well, for
historical reasons - I once was small and stupid ;)
About multiple domains, that is my target for sure.
I think the domain(in sip.conf) thing should come into help here,
where I associate a domain name to a context. I did try it once,
worked fine for a couple of test domains. But it seems I can't
associate one domain name to multple contexts. Am I correct?
You can specify one context for every domain your asterisk supports. On
one of my machines, a sip.conf might look like
8 sip.conf
[general]
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
domain=main.example.com,sip-in-examplecom
domain=private.example.org,sip-in-privateexampleorg
domain=customer.example.net,sip-in-customerexamplenet
8
So calls coming in for [EMAIL PROTECTED] are going through the
sip.conf context sip-in-examplecom.
In extensions.conf, I would configure like this:
8 extensions.conf
[sip-in-domains]
exten=_...,1,Set(A=${DB(callroute/names/[EMAIL PROTECTED])})
exten=_...,2,GotoIf($[A = A${A}]?900)
exten=_...,3,Goto(localdialplan,${A},1)
[sip-in-examplecom]
exten=_...,1,Set(DOMAIN=example.com)
exten=_...,2,Goto(sip-in-domains,${EXTEN},1)
[sip-in-privateexampleorg]
exten=_...,1,Set(DOMAIN=private.example.org)
exten=_...,2,Goto(sip-in-domains,${EXTEN},1)
[sip-in-customerexamplenet]
exten=_...,1,Set(DOMAIN=customer.example.net)
exten=_...,2,Goto(sip-in-domains,${EXTEN},1)
8
This would require database entries for users like
callroute/names/[EMAIL PROTECTED] = 201
callroute/names/[EMAIL PROTECTED] = 661
You can also have several domains map to the same users, e.g. you want
example.com and main.example.com to be equivalent, so you just add
another domain line to sip.conf, like
domain=example.com,sip-in-examplecom
You should be able to get around this multiple-context setup by using
the variable ${SIPDOMAIN} and only one context, but this somehow did not
work for me, so I came up with this solution. Play around, see if you
get it running. For me, it has been like this for a while, and then, I
try to avoid changing a running system. You could, for example, set all
your domains to
domain=example.net,sip-in-domains
and use
exten=_...,1,Set(A=${DB(callroute/names/[EMAIL PROTECTED])})
which _should_ work just as well.
You probably already found out that SRV records should be set for the
domains that asterisk is going to handle, let me give an example:
[EMAIL PROTECTED]:~$ dig @localhost example.org any
; (1 server found)
;; global options: printcmd
;; Got answer:
;; -HEADER- opcode: QUERY, status: NOERROR, id: 52979
;; flags: qr aa rd; QUERY: 1, ANSWER: 6, AUTHORITY: 0, ADDITIONAL: 1
;; WARNING: recursion requested but not available
;; QUESTION SECTION:
;example.org.IN ANY
;; ANSWER SECTION:
example-org. 604800 IN SOA ns1.example.net. root.example.org.
2007060504 21600 3600 1209600 21600
example.org. 604800 IN TXT v=spf1 mx a:mxs.example.org -all
example.org. 604800 IN MX 10 example.org.
example.org. 604800 IN A 81.12.999.999
example.org. 604800 IN NS ns1.example.net.
example.org. 604800 IN NS al25b.xi.yu.fiber.example.com.
example.org. 604800 IN NAPTR 60 50 s SIP+D2U
_sip._udp.example.org.
;; ADDITIONAL SECTION:
ns1.example.net.604800 IN A 81.12.999.999
;; Query time: 5 msec
;; SERVER: 127.0.0.1#53(127.0.0.1)
;; WHEN: Sat Sep 15 11:38:14 2007
;; MSG SIZE rcvd: 269
Where
_sip._udp.example.org. 604800 IN SRV 10 10 5060 example.org.
This is a setup with all web, mail and sip running on the same machine
(IP addresses and domains changed, of course) - but you should be able
to move things around so that those services actually can be run on
different machines.
Anything other to be done on Asterisk to support multiple domains?
Well, I think that is about enough ;-)
BR
Anselm
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Re: [asterisk-users] stop log/debug messages into /var/log/messages
Naa Bilal, haven't got to investigate it thoroughly yet. Kinda been occupied. Will let you know, if I do manage to do that. bilal ghayyad wrote: Dear Benjamin; OK friend, things are clear. But now I came to the same original issue that you asked about it, which is the ability to stop the log/debug messages into /var/log/messages. Same like your situation, the messages is comment (;) and even the logges are written to the /var/log/messages, so why that is happening? Did u find answer for that? Regards Bilal --- Benjamin Jacob [EMAIL PROTECTED] wrote: Hello Bilal, You have to do quite some reading mate, before you post your questions(like your nat and canreinvite questions). Anyway, look into /etc/asterisk/manager.conf for the required directories where Asterisk stores its various files/directories. Then read up logger.conf and look at some examples on the net as well. cheerz - Ben. bilal ghayyad wrote: Hi Benjamin; I am also interested in the same issue, but I would like to know how you can know where these logs are stored (in which file and path)? I readed that syslog, can you please help me about that? Regards Bilal Ghayad Mobile: 00965 9849460 --- When you access the A*k console, is this via a tty connection (ssh/telnet), or actually on the physical console of the server? I don't think it's A*k that's directly logging to the console - the config doesn't show that... I'm guessing, that you're accessing A*k via the local terminal, and that your syslog config for the server is configured to log this to messsages Maybe.. hmmm. interesting. need to investigate syslog now. Even me thinks, as far as I've read(abt logger and the existing configuration), it shouldn't be writing to any syslogs. btw, am accessing the * console via ssh. thanks for ur help. - Benjamin Jacob. Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit
Re: [asterisk-users] nat=yes
C F, I have nat=yes set by default for all my extensions(with canreinvite=no). And things work fine. Bilal, about Asterisk sending packets to public/private : Asterisk will send packets to the public IP advertised by the msg/recv from address. It is the NAT's headache on the endpoints network periphery to send the response from Asterisk to the endpoint. C F wrote: If you set yes then asterisk assumes that the address its coming from is not the same as the UA thinks it is. most devices will not operate properly if set to yes when they are in fact local. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] stop log/debug messages into /var/log/messages
Adrian Marsh wrote: When you access the A*k console, is this via a tty connection (ssh/telnet), or actually on the physical console of the server? I don't think it's A*k that's directly logging to the console - the config doesn't show that... I'm guessing, that you're accessing A*k via the local terminal, and that your syslog config for the server is configured to log this to messsages Maybe.. hmmm. interesting. need to investigate syslog now. Even me thinks, as far as I've read(abt logger and the existing configuration), it shouldn't be writing to any syslogs. btw, am accessing the * console via ssh. thanks for ur help. - Benjamin Jacob. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Jacob Sent: 04 September 2007 12:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] stop log/debug messages into /var/log/messages Here it is : SIP01*CLI logger show channels Channel Type StatusConfiguration --- --- Console Enabled- Notice Error EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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
[asterisk-users] alphabetical extension patterns
Hello ppl, Any way to specify alphabetical exten patterns in the dialplans on Asterisk? All my users would have alpha/numerical ids. I don't want to add a line for every user in my dialplans. I searched around, but couldn't get anything useful. Any way to get around this? Thanks in advance - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
[asterisk-users] stop log/debug messages into /var/log/messages
Hello good ppl, Any way of stopping asterisk writing into syslogs or any other file, if I set verbose 6 on the console? All I want is the verbose output only on the console, nowhere else. My logger.conf says : console= notice,error ;messages = notice,warning,error Thanks in advance. - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] stop log/debug messages into /var/log/messages
Btw, even the syslog line in logger.conf is commented : ; syslog.local0 = notice,warning,error Benjamin Jacob wrote: Hello good ppl, Any way of stopping asterisk writing into syslogs or any other file, if I set verbose 6 on the console? All I want is the verbose output only on the console, nowhere else. My logger.conf says : console= notice,error ;messages = notice,warning,error Thanks in advance. - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] stop log/debug messages into /var/log/messages
Exactly the same lines as on the console. Adrian Marsh wrote: What logs are coming out to /var/log/messages? Adrian Marsh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Jacob Sent: 04 September 2007 07:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] stop log/debug messages into /var/log/messages Btw, even the syslog line in logger.conf is commented : ; syslog.local0 = notice,warning,error Benjamin Jacob wrote: Hello good ppl, Any way of stopping asterisk writing into syslogs or any other file, if I set verbose 6 on the console? All I want is the verbose output only on the console, nowhere else. My logger.conf says : console= notice,error ;messages = notice,warning,error Thanks in advance. - Benjamin Jacob. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit
Re: [asterisk-users] stop log/debug messages into /var/log/messages
Here it is : SIP01*CLI logger show channels Channel Type StatusConfiguration --- --- Console Enabled- Notice Error Tzafrir Cohen wrote: On Tue, Sep 04, 2007 at 10:43:15AM +0100, Adrian Marsh wrote: What logs are coming out to /var/log/messages? Ask asterisk logger show channels EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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
[asterisk-users] [Fwd: Re: issues with caller ID , remote-party-id
Hello ppl,
Sorry to re-post it, but kinda these issues are getting on my nerves.
I tried Set(CALLERID(num)=7329) on 1.2.12, which works fine, but not on
1.4.4.
The problem :
1. I receive call from caller 'AAA' on my number, 'BBB' which is on my
Asterisk box.
2. I have to redirect the call to some other number, say, my cell num -
'CCC'.
3. My PSTN provider wants the calling(From) number as 'BBB', which is
fair enough, because that number has been assigned to me by this provider.
4. I have been able to achieve this(using Set(CALLERD(num)='BBB'), on
1.2.12, but not on 1.4.4. I know, be default, From will be set to BBB,
but still
5. But, more importantly, I need to pass the original caller number too
to the destination, i.e. to my cell fone - CCC, which shows up on my
cell fone as the caller id.
6. I presume, this can be achieved using Remote-Party-ID.
7. If I set sendrpid=yes in sip.conf, the stuff sent in Remote-Party-ID
also is CCC, but I want it to be AAA (actual caller).
8. So, I commented out sendrpid, and manually added Remote-Party-ID using:
SIPAddHeader(Remote-Party-ID: MEUSER AAA\;privacy=off\;screen=no)
9. As I am experimenting, I don't really have PSTN connectivity yet, but
I have come to know, that most devices, like CISCO 7960, give preference
to Remote-Party-ID over the From number to show as Caller ID. So, I have
CCC configured on a CISCO SIP phone. But the caller id is still 'BBB'.
10. And at the end of all this, I am very close to smash my asterisk
box, cisco phones with a sledgehammer.
Any bright ideas anywhere???
Help appreciated.
Thanks..
- Ben.
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---BeginMessage---
Also,
. if I use Remote-party-id header, can it be different from the 'From' URI?
. If yes, how do you achieve this in Asterisk?
. What(From or Remote-party-id) is used by clients to show as the CLI of
the caller?
if I am not mistaken, Remote-party-id is for network elements to confirm
identities of end subscribers.
All corrections and suggestions welcome.
- Ben
Benjamin Jacob wrote:
Hello All,
Is CALLERID() setting broken in 1.4.4?
My small dialplan :
[testclid]
exten = _0.,1,Set(CALLERID(all)=Ben Jacob 988077)
exten = _0.,n,Dial(SIP/${EXTEN})
Correct me if I am wrong, Set(CALLERID(all) above supposed to change the
display name as above(Ben Jacob) and change the From URI to [EMAIL PROTECTED]
As of now, only the _display name_ is being replaced, but not the name.
I tried CALLERID(num) as well CALLERID(number), to the same effect(only
display name being set to number).
Anyone facing similar problems?
Thanks in advance.
- Ben
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[asterisk-users] 1.4.4. caller ID not working ?
Hello All,
Is CALLERID() setting broken in 1.4.4?
My small dialplan :
[testclid]
exten = _0.,1,Set(CALLERID(all)=Ben Jacob 988077)
exten = _0.,n,Dial(SIP/${EXTEN})
Correct me if I am wrong, Set(CALLERID(all) above supposed to change the
display name as above(Ben Jacob) and change the From URI to [EMAIL PROTECTED]
As of now, only the _display name_ is being replaced, but not the name.
I tried CALLERID(num) as well CALLERID(number), to the same effect(only
display name being set to number).
Anyone facing similar problems?
Thanks in advance.
- Ben
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by reply email and then destroy the message. Opinions, conclusions and other
information in this message that do not relate to official business of Mascon
shall be understood to be neither given nor endorsed by Mascon. Any information
contained in this email, when addressed to Mascon clients is subject to the
terms and conditions in governing client contract.
Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we
can not guarantee that any email or attachment is free from computer viruses
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Re: [asterisk-users] 1.4.4. caller ID not working ?
Also,
. if I use Remote-party-id header, can it be different from the 'From' URI?
. If yes, how do you achieve this in Asterisk?
. What(From or Remote-party-id) is used by clients to show as the CLI of
the caller?
if I am not mistaken, Remote-party-id is for network elements to confirm
identities of end subscribers.
All corrections and suggestions welcome.
- Ben
Benjamin Jacob wrote:
Hello All,
Is CALLERID() setting broken in 1.4.4?
My small dialplan :
[testclid]
exten = _0.,1,Set(CALLERID(all)=Ben Jacob 988077)
exten = _0.,n,Dial(SIP/${EXTEN})
Correct me if I am wrong, Set(CALLERID(all) above supposed to change the
display name as above(Ben Jacob) and change the From URI to [EMAIL PROTECTED]
As of now, only the _display name_ is being replaced, but not the name.
I tried CALLERID(num) as well CALLERID(number), to the same effect(only
display name being set to number).
Anyone facing similar problems?
Thanks in advance.
- Ben
EMAIL DISCLAIMER : This email and any files transmitted with it are
confidential and intended solely for the use of the individual or entity to
whom they are addressed. Any unauthorised distribution or copying is strictly
prohibited. If you receive this transmission in error, please notify the
sender by reply email and then destroy the message. Opinions, conclusions and
other information in this message that do not relate to official business of
Mascon shall be understood to be neither given nor endorsed by Mascon. Any
information contained in this email, when addressed to Mascon clients is
subject to the terms and conditions in governing client contract.
Whilst Mascon takes steps to prevent the transmission of viruses via e-mail,
we can not guarantee that any email or attachment is free from computer
viruses and you are strongly advised to undertake your own anti-virus
precautions. Mascon grants no warranties regarding performance, use or quality
of any e-mail or attachment and undertakes no liability for loss or damage,
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shall be understood to be neither given nor endorsed by Mascon. Any information
contained in this email, when addressed to Mascon clients is subject to the
terms and conditions in governing client contract.
Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we
can not guarantee that any email or attachment is free from computer viruses
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Re: [asterisk-users] multiple pbxes, multiple domains, same user ids?
Anthony Francis wrote: Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the domain field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with the same specified in sip.conf). But, Asterisk maintains the registration of the latest registree!! thats really sad for me . Any work around for this one(multiple pbx)? I would be zapped and amazed if multiple pbx isn't possible in Asterisk. Help anyone? cheers - Ben. you have to do different sip-ids, I am guessing you are probably using the extension #, you dont need to do that. What do you mean by multiple-pbx's anyway? I hope you don't mean multiple instances of *.What I am sure you mean is multiple dial plans, and yes, * is multi-tenant friendly. What we do for uniqueness is use the last 8 digits of the device mac addr or other unique number followed by a dash - followed by the extension number. Anthony Thanks Anthony. I definitely don't mean multiple instances of asterisk. Multiple dial plans, hmm.. yes.. in a way. Multiple pbx ... in short, provide pbxes for two entirely different organizations, say, Microsoft and IBM (can i use these names in here? ;-) ). Each would have many extensions, but each office can have identical extensions, e.g. you can have extensions 4001 in both. But one would be [EMAIL PROTECTED] and the other would be [EMAIL PROTECTED] . [EMAIL PROTECTED] should be able to call any user within Microsoft. To step outside the organization, you would put in some logic(dialplans). So, i want to have pbx for microsoft and another pbx for IBM. Is it possible to have two or more pbxes within one Asterisk instance. Hope you got my point. cheerz - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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
[asterisk-users] multiple pbxes, multiple domains, same user ids?
Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the domain field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with the same specified in sip.conf). But, Asterisk maintains the registration of the latest registree!! thats really sad for me . Any work around for this one(multiple pbx)? I would be zapped and amazed if multiple pbx isn't possible in Asterisk. Help anyone? cheers - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] multiple pbxes, multiple domains, same user ids?
Ouch. And I thought I had an answer to my query. I totaly agree abt the long disclaimer nonsense Schmaltz, but I swear by the powers up there, it's the admins over here at my workplace doing all that nonsensical magic, as the mails go out. I wish i had the freedom to use gmail(just like you), thru the day, and not the office mail servers! Do you have any idea as to how do I get rid of this disclaimer whenever I mail to the Asterisk Users mailing list?? Pray, tell me! Btw, did you happen to read my query, or you straight on jumped to the disclaimer? roving eyes, eh? Any answers anyone , to my query(abt multiple pbxes)? Apologies if I am missing something elementary here. cheerz - Ben. C F wrote: Can you please get rid of your awfull long nonsense disclaimer? On 8/1/07, Benjamin Jacob [EMAIL PROTECTED] wrote: Hello good ppl, A couple of questions for multiple pbxes 1. Is it possible to support multiple pbxes in one Asterisk box(using contexts, etc.)? 2. Can we use the domain field in sip.conf to specify the different domains for sip users, having one domain for each pbx? I just tried registering two xlites, with different domain names (with the same specified in sip.conf). But, Asterisk maintains the registration of the latest registree!! thats really sad for me . Any work around for this one(multiple pbx)? I would be zapped and amazed if multiple pbx isn't possible in Asterisk. Help anyone? cheers - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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
[asterisk-users] asterisk on 64-bit?
Hello ppl, Searched all over, but couldn't find anything conclusive. Does an off-the-shelf version of Asterisk run without any issues on a 64-bit machine? Does anyone have any 'conclusive' figures? Apologies if this is a repeat question. Would appreciate if I could be redirected to the appropriate link. cheerz - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] They ignore my DTMF!
Pierre, Thats exactly what Joanna said in her reply. Check the client DTMF settings on your phones. set it to rfc2833 or out-of-band, whatever the config says. Grandstream by default have inband DTMF set, and usualy ulaw is supported as well, and thats the reason ur grandstream works but others dont. cheerz - Ben. Pierre Marceau wrote: Hi Joanna, Thanks for your reply. In my mind I think it must be some setting in the client (phone) becasue the Grandstream GXP 2000 does work and it is using the same sip.conf Extensions: 6000 is xlite softfone 6003 is Linksys SPA941 6004 is Grandstream GXP 2000 6005 is Linksys PAP2NA Please have a look at my sip conf and suggest any changes I could try... [general] context=internal bindport=5060 bindaddr=0.0.0.0 srvlookup=yes type=friend secret=XXX nat=no host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw subscribecontext=internal canreinvite=no register=8885551234:[EMAIL PROTECTED] [atlasvoice] type=friend host=proxy.atlasvoice.com username=8885551234 secret=XXX fromuser=8885551234 fromdomain=proxy.atlasvoice.com canreinvite=no insecure=very nat=yes context=incoming [6000] [EMAIL PROTECTED] [6001] [6003] [6004] [6005] [6006] [6007] [6008] Thanks, Pierre [EMAIL PROTECTED] 2/20/2007 10:47 PM Hi Pierre, Just a thought..check your dtmfmode in your SIP client configuration, if your using inband but your codec is not ulaw or alaw the DTMF tones will be misrepresented and thus will not be recognised due to the audio compression, on the other hand if your phones are rfc2833 and asterisk is set to inband you wont hear anything. Hope that helps. Best Regards, Joanna On 2/21/07, Pierre Marceau [EMAIL PROTECTED] wrote: Hello, I can call out to the PSTN and talk to people but when I have to enter a dtmf tone in an ivr or voicemail system those systems do not recognise that I have sent a tone. This is the case when I make the call with the Xlite softfone or a regular telephone plugged into a PAP2NA or a Linksys SPA941. However... a Grandstream GXP2000 works just great ??? All are extensions on my Asterisk 1.4 box. I am using a voip trunk through Atlasvoice. All extensions are setup identical in sip.conf. One last thing, if a system wants me to respond 1 for sales 2 for service I can hit the 1 button quickly 4 or 5 times and the remote system will get it. That does not work for a three digit extension as you may well imagine. Any help would be appreciated. Pierre ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- The problem with the Future is that it keeps turning into the Present. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] They ignore my DTMF!
rfc2833 is the prefered way, as inband will work perfectly only with the ulaw codec. Pierre Marceau wrote: Okay, in the SPA-941 admin I changed: ;DTMF Tx Method: Auto DTMF Tx Method: Inband and now it works. Thanks! Pierre [EMAIL PROTECTED] 2/21/2007 12:09 AM Pierre, Thats exactly what Joanna said in her reply. Check the client DTMF settings on your phones. set it to rfc2833 or out-of-band, whatever the config says. Grandstream by default have inband DTMF set, and usualy ulaw is supported as well, and thats the reason ur grandstream works but others dont. cheerz - Ben. Pierre Marceau wrote: Hi Joanna, Thanks for your reply. In my mind I think it must be some setting in the client (phone) becasue the Grandstream GXP 2000 does work and it is using the same sip.conf Extensions: 6000 is xlite softfone 6003 is Linksys SPA941 6004 is Grandstream GXP 2000 6005 is Linksys PAP2NA Please have a look at my sip conf and suggest any changes I could try... [general] context=internal bindport=5060 bindaddr=0.0.0.0 srvlookup=yes type=friend secret=XXX nat=no host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw subscribecontext=internal canreinvite=no register=8885551234:[EMAIL PROTECTED] [atlasvoice] type=friend host=proxy.atlasvoice.com username=8885551234 secret=XXX fromuser=8885551234 fromdomain=proxy.atlasvoice.com canreinvite=no insecure=very nat=yes context=incoming [6000] [EMAIL PROTECTED] [6001] [6003] [6004] [6005] [6006] [6007] [6008] Thanks, Pierre [EMAIL PROTECTED] 2/20/2007 10:47 PM Hi Pierre, Just a thought..check your dtmfmode in your SIP client configuration, if your using inband but your codec is not ulaw or alaw the DTMF tones will be misrepresented and thus will not be recognised due to the audio compression, on the other hand if your phones are rfc2833 and asterisk is set to inband you wont hear anything. Hope that helps. Best Regards, Joanna On 2/21/07, Pierre Marceau [EMAIL PROTECTED] wrote: Hello, I can call out to the PSTN and talk to people but when I have to enter a dtmf tone in an ivr or voicemail system those systems do not recognise that I have sent a tone. This is the case when I make the call with the Xlite softfone or a regular telephone plugged into a PAP2NA or a Linksys SPA941. However... a Grandstream GXP2000 works just great ??? All are extensions on my Asterisk 1.4 box. I am using a voip trunk through Atlasvoice. All extensions are setup identical in sip.conf. One last thing, if a system wants me to respond 1 for sales 2 for service I can hit the 1 button quickly 4 or 5 times and the remote system will get it. That does not work for a three digit extension as you may well imagine. Any help would be appreciated. Pierre ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- The problem with the Future is that it keeps turning into the Present. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] s-${DIALSTATUS} extensions
Make it
Goto(s-${DIALSTATUS})
cheerz
- Ben.
Yuan LIU wrote:
In examples, s-${DIALSTATUS} is used to handle unsuccessful dial
attempts in the s extension. Goto() is used in examples. Is the
prefix s- mandatory? Is it related to the original extension s?
(Apparently Goto(${DIALSTATUS}) won't work for me.)
Yuan Liu
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Re: [asterisk-users] Polycom IP 501+India
If you already havent seen this: http://dir.indiamart.com/impcat/video-telephone.html cheerz - Ben. Crazy Boy wrote: Hi Friends, This is Chandra from India. I have installed and configured Asterisk in our company. I want to provide Polycom IP 501 model phones to our employees. I am unable to find the dealer for these phones in India. Where can I buy these phones in India? If anybody knows, please tell me the dealer address or phone number. This is very urgent. Looking forward to your response. Thank you. Regards, Chandra. Everyone is raving about the all-new Yahoo! Mail beta. http://us.rd.yahoo.com/evt=45083/*http://advision.webevents.yahoo.com/mailbeta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- The problem with the Future is that it keeps turning into the Present. *** EMAIL DISCLAIMER : *** This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ** ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP directly
Davida, You would also want to look at canreinvite option in sip.conf http://www.voip-info.org/wiki-Asterisk+SIP+canreinvite cheerz - Ben. Eric ManxPower Wieling wrote: David Alcott wrote: Is there a way to configure the Asterisk so that the RTP goes directly between the Endpoints as opposed to going through the asterisk? That is the default if Asterisk believes it will work. Things that might not make it work is tTwW options to Dial, protocol transation (one leg is SIP, the other is IAX2, transcoding, NAT, or many other things that make the two legs of the call not compatible with reinvites. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users *** EMAIL DISCLAIMER : *** This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ** ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: CLI History
And ofcourz, be careful, with your fingers on the CLI or elsewhere, esp on a production server. cheerz - Ben. Benny Amorsen wrote: DG == Douglas Garstang [EMAIL PROTECTED] writes: DG When I exited the CLI and re-entered and pressed ctrl-c, That's where your problem is. Use exit and not ctrl-c to leave asterisk -r. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller ID authentication
Use astdb for such apps. Look at Lookupblacklist, similarly, you can set up ur whitelist http://www.asteriskguru.com/tutorials/lookupblacklist.html Vernier Umali wrote: I looked at the ex-girlfriend option and it's just part of what I needed. What I do want is to setup a whitelist or numbers which can access the asterisk box and its extensions. All other numbers will be given a congestion or busy tone regardless of what extension they are trying to reach. It would be better that the whitelist is in an external database of list that asterisk can look up. On 12/13/06, Vernier Umali [EMAIL PROTECTED] wrote: Thanks On 12/13/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Vernier Umali wrote: Is there a utility or srcipt in asterisk which accepts calls based on caller ID and gives a busy signal if the caller ID is not on the list. Thanks Search the Wiki or Mailing List archives for the ex-girlfriend option. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI History
On Mon, 2006-12-11 at 10:31 -0700, Douglas Garstang wrote: What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, the last command in the history always defaults to 'stop now'. This is very bad, and it's caused accidental shutdowns more than once. thats prety smart... think hard.. wot was the command u gave to exit the CLI?? history is a last-in-first-out kinda setup, anywhere, not just in * CLI. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mwi for voicemail not showing up for realtimeconfig.
Got it mate. thanx for that. Am using mysql for voicemail storage, unlike in the script you've written which works on mails on disk on a certain path. All I've to do is query for INBOX(new) and Old(old) voicemessages count. cheerz - Ben Scott Keagy wrote: A while back I posted a fully functional though somewhat elaborate mechanism to get MWI working with real-time voicemail and NOT using static (static kinda takes a big chunk of value away from real-time). Search the digium Asterisk User forums for my username skeagy with keyword mwi. It does not rely on the built-in sip mechanism. It's a system of scripts that are either triggered by asterisk or a cron-job every one minute to clean out a spool directory, and it uses a uses a template SIP message in a file along with sipsak. It's been working 100% flawlessly in production for 11 months. I'm sure it would work with Asterisk 1.4beta3 assuming that voicemail.conf can still trigger an external script. Regards, Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber Sent: Monday, December 04, 2006 4:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. Since I started using 1.4 I'm also not getting MWI. I am not using realtime. MARK. Benjamin Jacob wrote: Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the db. Any ideas, wot else shud i check for? TiA. cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mwi for voicemail not showing up for realtime config.
Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the db. Any ideas, wot else shud i check for? TiA. cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] seed vs registration?
Hello ppl, The scenario : I restart asterisk, sip show peers shows nothing. I make a call from 7013 to 7011. I get the following o/p : SIP Seeding peer from astdb: '7013' at [EMAIL PROTECTED] for 3600 SIP Seeding peer from astdb: '7011' at [EMAIL PROTECTED] for 240 And then the call goes thru. So, does 'Seeding', means * registers both users?? But a subsequent REGISTER msg shows : Registered SIP '7011' at 192.168.10.53 port 10016 expires 240 n so on. So how does REGISTER differ from Seeding? Also, what should be the defined behaviour if the caller and/or callee is/are not registered when they attempt a call(INVITE)? Thanks in advance. cheerz - Ben ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail, SQL ODBC
AFAIK, ODBC helps you have any DB underneath, be it MySQL, PGSQL, etc., so why not go ahead with it? cheerz - Ben. Norbert Zawodsky wrote: Hi Peder, I asked the same question some time ago. Never got any answer... :-( Norbert Peder @ NetworkOblivion schrieb: Is the storage of actual voicemail messages in a database still limited to ODBC? If so, why? And is the use of mySQL and ODBC at the same time still a bad idea? If so, why? I want to store all of my voicemail stuff in a database so that I can give users web access to it, but I don't want to run web services on my * server itself. If it is all in a DB, I can have a web box and a separate SQL box and none of it should affect *. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 3xx redirect from asterisk?
Hello ppl, Is it possible to send a REDIRECT from an Asterisk box, to an incoming call?? e.g. A calling B, via Asterisk, Asterisk sends redirect to A to contact C. cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time Based Voicemail Messages
a lil bit of googling wud have answered you Tim. Put in some effort next time anyway, for now : http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+openhours Wildheart wrote: Hi, I want to change my voicemail message based on the time of day. I would like a message that says Sorry the office is now closed. after a certain time, and says Sorry I am unavailable / Busy / etc before. I have come up with two ways of doing it: 1. A cron job to replace the files (messy) 2. Using different mailboxes at the different times (this means I have 2 mailboxes to check). Is there a way that the voicemail could be enhanced by adding a feature like this? With thanks, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] faxing times!
Hello ppl, Reading all over the net. Learnt quite a lot, but that has left me confused-a-lot as well. Need answers to a few questions. Before that, I have an ISP(fax gateway) which will help me send/recv faxes using the T.38 protocol. I am using Asterisk 1.2.12.1. Now to the few questions I had: 1) Do I need any additional hardware on the Asterisk box?? I did download the spandsp and rxfax and txfax, n email2fax packages. But it seems, all those work on the Zap channels. 2) So far, I've worked ONLY with SIP and IAX. So, is it possible to do fax-ing over these? How? Help!! - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mapping CLI'S in Dialplan
Your offnet calls will be more than 4 digits, so use that to ur advantage.
so, for internal calls,
exten = _,1,Set(CALLERID(all)=Name ${CALLERID(num)})
or if u dont want to change the CLID at all.. dont do anything..
exten = _,1,NoOp(nothing)
else, for all external calls(4 digits)
exten = _X.,1,Set(CALLERID(num)=urDID)
cheerz
- Ben.
Scott Pinhorne wrote:
Hi All
I am not sure what I wish to do it possible but I would like to see if
you guys know any better.
I have a site who has the extensions: 1231, 1232. 1233, 1234
Each of these users can dial each other on the extension number an
also has an external CLI mapped to them.
On all internal calls or calls to services such as call forwarding
their Caller ID is: Name
What I would like to have happen is have the Caller ID changed to the
CLI only when they make an offnet call.
So what I am saying is I need to match an extension to a CLI and reset
the Caller ID.
Many Thanks
SP
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Re: SV: [asterisk-users] ip address in CDR
Just the answer I expected. But, how do I get the IPs of the two parties? Jon Schøpzinsky wrote: You can use the CDR(userfield) value, to save the ip's in the CDR record. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Benjamin Jacob Sendt: 3. november 2006 06:18 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [asterisk-users] ip address in CDR Hello ppl, Any way to store the origination or termination IP addresses in CDRs? cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ip address in CDR
Hello ppl, Any way to store the origination or termination IP addresses in CDRs? cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk n QoS
I know, I know, the wiki link for that one. But wot I wanted were actual figures related to Asterisk n QoS. How does Asterisk actualy handle and fare at the following QoS issues : 1) Delay 2) Jitter 3) Packet loss These and more ideas are welcome. cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Fwd: Asterisk n QoS]
Not too sure, if this msg did reach the group, so resending. ---BeginMessage--- I know, I know, the wiki link for that one. But wot I wanted were actual figures related to Asterisk n QoS. How does Asterisk actualy handle and fare at the following QoS issues : 1) Delay 2) Jitter 3) Packet loss These and more ideas are welcome. cheerz - Ben. ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: IAX2 goes one way audio when lag gets bad
Martin Joseph wrote: On 2006-10-25 08:14:43 -0700, Noah Miller [EMAIL PROTECTED] said: Hi Matt - I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. On the customer's end I have the following config in iax.conf: trunk=no (I have also tried trunk=yes and nothing for trunk=) jitterbuffer=yes forcejitterbuffer=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 If you're using Asterisk 1.2.x, dropcount, jittershrinkrate and minexcesbuffer don't do anything. They are ignored by 1.2.x unless you specify that you want to use the old 1.0.x jitterbuffer. Instead you might try the parameters maxjitterbuffer, resyncthreshold, and maxjitterinterps. For more, you can check out the sample iax.conf. I believe, also, that you are correct in setting trunk=no. I know in the 1.0.x jitterbuffer, trunk was not fully supported. I think this is still the case with the 1.2.x jitterbuffer. If the audio is dropping out completely, then I suspect the whole jitter buffer thing is a red herring (waste of time). Perhaps it's a nat issue? What kind of router if any is involved? I am reaching here... Also, please do tell us which version of asterisk you are running... Marty seeing this thread a lil too late, i guess. So, am sorry if I am repeating things. When I was setting up my iax2 configs, I too had one way audio initialy. Tried the softphone on two machines(which incidentaly had asterisk running on them as well), to no avail. When I looked at the tcpdump on my asterisk server, I could see no rtp coming in from the two said machines. So, I shifted the softphone to another machine, this time on a windows machine, n voila! it worked like a charm. So, I hope you did have a look at the tcpdump to check on the rtp flow. cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail maintenance
Arnd Vehling wrote: Jordan Novak wrote: Has anyone created a GUI for this. I am not sure what youre looking for but we developed a Voicemail Manager: = http://sip-syndication.com best regards, Arnd Hello Vehling, This product of yours, does it manipulate, files on the Asterisk server itself? If yes, does that mean, this has to be installed on the same server as Asterisk? As for you, Jordan, you can very easily create GUIs for voicemail management, if you store your voicemails in sql db. www.voip-info.org/wiki/view/*Asterisk*+Voicemail+*ODBC*+*storage . cheerz - Ben. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime... Help Me!!!
Alls well that ends well !!! :-) Maurizio Pederneschi wrote: Great! Thanks for your aid... I spend a lot of day around this problem... Now realtime load returns data! - Original Message - From: Tijl Van den Broeck [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 23, 2006 7:52 AM Subject: Re: [asterisk-users] Asterisk Realtime... Help Me!!! Thanks alot! Indeed it was the 3th solution, I changed sipusers = odbc,MySQL-asterisk,sip_buddies sippeers = odbc,MySQL-asterisk,sip_buddies into sipusers = odbc,mysql2,sip_buddies sippeers = odbc,mysql2,sip_buddies And realtime load sipusers username 1006 now returns data :-) greets Tijl Van den Broeck On 10/23/06, Benjamin Jacob [EMAIL PROTECTED] wrote: Make additional checks : 1) ensure u've unixodbc, unixodbc-devel installed, use this command rpm -qa | grep -i unixodbc MUST see unixodbc and unixodbc-devel in the output!!!, else get unixodbc and unixodbc-devel(am kinda guessing u do have that perfect). 2) /etc/odbcinst.ini and /etc/odbc.ini should be correct. cross check 3) Aaahh.. revelation!! I think, I know where you've gone wrong. In your res_odbc.conf , you have given the database context as mysql(see [mysql]). This should be the same as the 2nd argument in ur extconfig.conf line for realtime for your sipusers. i.e. it should be sipusers = odbc,mysql,sipusers instead of sipusers = odbc,asterisk,sipusers This should work fine. If it doesn't, paste your odbc.ini and odbcinst.ini files as well over here. or give me ssh login access to your machine.(dont wory, wont mess up ur machine). cheerz - Ben. Maurizio Pederneschi wrote: These are my conf file: res_odbc.conf ;;; odbc setup file ; ENV is a global set of environmental variables that will get set. ; Note that all environmental variables can be seen by all connections, ; so you can't have different values for different connections. [ENV] INFORMIXSERVER = my_special_database INFORMIXDIR = /opt/informix ; All other sections are arbitrary names for database connections. ;[asterisk] ;enabled = yes ;dsn = asterisk ;;username = myuser ;;password = mypass ;pre-connect = yes [mysql] enabled = yes dsn = MySQL-asterisk username = root password = pre-connect = yes --- - - extconfig.conf ; ; Static and realtime external configuration ; engine configuration ; ; Please read doc/README.extconfig for basic table ; formatting information. ; [settings] ; ; Static configuration files: ; ; file.conf = driver,database[,table] ; ; maps a particular configuration file to the given ; database driver, database and table (or uses the ; name of the file as the table if not specified) ; ;uncomment to load queues.conf via the odbc engine. ; ;queues.conf = odbc,asterisk,ast_config ; ; The following files CANNOT be loaded from Realtime storage: ; asterisk.conf ; extconfig.conf (this file) ; logger.conf ; ; Additionally, the following files cannot be loaded from ; Realtime storage unless the storage driver is loaded ; early using 'preload' statements in modules.conf: ; manager.conf ; cdr.conf ; rtp.conf ; ; ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; ;example = odbc,asterisk,alttable ;iaxusers = odbc,asterisk ;iaxpeers = odbc,asterisk sipusers = odbc,asterisk,sipusers ;sippeers = odbc,asterisk voicemail = odbc,asterisk ;extensions = odbc,asterisk ;queues = odbc,asterisk ;queue_members = odbc,asterisk extensions = odbc,asterisk,extensions --- - This is my table sipusers | id | name | username | context | host| port | secret | allow | ipaddr | type | password | | 1 | pippo| pippo| tutorial | dynamic | | password | g729;ilbc;gsm;ulaw;alaw | NULL | friend | password | | 2 | testAsterisk | testAsterisk | tutorial | dynamic | | password | g729;ilbc;gsm;ulaw;alaw | NULL | friend | password | --- - This is the output of the realtime load command: realtime load sipusers name pippo No rows found matching search criteria. Thank's Maury - Original Message - From: Benjamin Jacob [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 20, 2006 12:39 PM Subject: Re: [asterisk-users] Asterisk Realtime... Help Me!!! Maurizio Pederneschi wrote: Hi, i have
Re: [asterisk-users] accountcode and amaflags?
Any more ideas, esp from guys whove used this in their setp? Benjamin Jacob wrote: Giovanni, Appreciate your lines mate. But, Ive already read those, all over the net. my qs inline : amaflags : Categorization for CDR records. Choices are default, omit, billing, documentation and choices are defaul, omit, billing, documentation wot r these categories??wot decides these categories? accountcode : string : Users may be associated with an accountcode (billing purpose) hmm.. ive seen in quite a few places, where the pin collected is stored as the accountcode... wot duz that mean? anyway, can you give me an example of wot the association means?am a lil slow.. Cheers, Giovanni 2006/10/19, Benjamin Jacob [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hello ppl, Can someone explain to me the meaning and use of the variables accountcode and amaflags in sip.conf,etc. Googled, voip-infoed, wikied, etc for it. Couldnt get much of it. I know, they are billing related, but not much beyond that. Any ideas? cheerz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime... Help Me!!!
Make additional checks : 1) ensure u've unixodbc, unixodbc-devel installed, use this command rpm -qa | grep -i unixodbc MUST see unixodbc and unixodbc-devel in the output!!!, else get unixodbc and unixodbc-devel(am kinda guessing u do have that perfect). 2) /etc/odbcinst.ini and /etc/odbc.ini should be correct. cross check 3) Aaahh.. revelation!! I think, I know where you've gone wrong. In your res_odbc.conf , you have given the database context as mysql(see [mysql]). This should be the same as the 2nd argument in ur extconfig.conf line for realtime for your sipusers. i.e. it should be sipusers = odbc,mysql,sipusers instead of sipusers = odbc,asterisk,sipusers This should work fine. If it doesn't, paste your odbc.ini and odbcinst.ini files as well over here. or give me ssh login access to your machine.(dont wory, wont mess up ur machine). cheerz - Ben. Maurizio Pederneschi wrote: These are my conf file: res_odbc.conf ;;; odbc setup file ; ENV is a global set of environmental variables that will get set. ; Note that all environmental variables can be seen by all connections, ; so you can't have different values for different connections. [ENV] INFORMIXSERVER = my_special_database INFORMIXDIR = /opt/informix ; All other sections are arbitrary names for database connections. ;[asterisk] ;enabled = yes ;dsn = asterisk ;;username = myuser ;;password = mypass ;pre-connect = yes [mysql] enabled = yes dsn = MySQL-asterisk username = root password = pre-connect = yes - extconfig.conf ; ; Static and realtime external configuration ; engine configuration ; ; Please read doc/README.extconfig for basic table ; formatting information. ; [settings] ; ; Static configuration files: ; ; file.conf = driver,database[,table] ; ; maps a particular configuration file to the given ; database driver, database and table (or uses the ; name of the file as the table if not specified) ; ;uncomment to load queues.conf via the odbc engine. ; ;queues.conf = odbc,asterisk,ast_config ; ; The following files CANNOT be loaded from Realtime storage: ; asterisk.conf ; extconfig.conf (this file) ; logger.conf ; ; Additionally, the following files cannot be loaded from ; Realtime storage unless the storage driver is loaded ; early using 'preload' statements in modules.conf: ; manager.conf ; cdr.conf ; rtp.conf ; ; ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; ;example = odbc,asterisk,alttable ;iaxusers = odbc,asterisk ;iaxpeers = odbc,asterisk sipusers = odbc,asterisk,sipusers ;sippeers = odbc,asterisk voicemail = odbc,asterisk ;extensions = odbc,asterisk ;queues = odbc,asterisk ;queue_members = odbc,asterisk extensions = odbc,asterisk,extensions This is my table sipusers | id | name | username | context | host| port | secret | allow | ipaddr | type | password | | 1 | pippo| pippo| tutorial | dynamic | | password | g729;ilbc;gsm;ulaw;alaw | NULL | friend | password | | 2 | testAsterisk | testAsterisk | tutorial | dynamic | | password | g729;ilbc;gsm;ulaw;alaw | NULL | friend | password | This is the output of the realtime load command: realtime load sipusers name pippo No rows found matching search criteria. Thank's Maury - Original Message - From: Benjamin Jacob [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 20, 2006 12:39 PM Subject: Re: [asterisk-users] Asterisk Realtime... Help Me!!! Maurizio Pederneschi wrote: Hi, i have implemented Asterisk Realtime architecture with Odbc and MySql DB. I have followed all the step of the documentation I found on the Internet. On the CLI, if I make odbc show I see that the DB connection is UP, but if I make realtime load family column value both with extensions family or with sipusers family, I can't find anything in the db. Why it happens? What can I check in my configuration? Someone know if there is a way to test if asterisk make effectively the query to the DB when I make the realtime load command? Please, help me! Maury ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users paste your relevant config files and also an example command (realtime load etc
Re: [asterisk-users] Re: checking 'voicemail externally - doesn't work
Avi Miller wrote: On 23/10/2006, at 2:35 PM, Eric ManxPower Wieling wrote: Works for me. 1.2.12.1 with FreePBX. When I press *, I get a password prompt. Entering my password gets me into the main voicemail menu. FreePBX is NOT Asterisk. Yes, I know that. Hence the 1.2.12.1 *with* FreePBX statement. I.E. Asterisk v1.2.12.1 *with* FreePBX *added* I know what FreePBX is. I also know the differences between Asterisk, FreePBX, [EMAIL PROTECTED] and TrixBox. :) Pray, tel me difference!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime... Help Me!!!
Maurizio Pederneschi wrote: Hi, i have implemented Asterisk Realtime architecture with Odbc and MySql DB. I have followed all the step of the documentation I found on the Internet. On the CLI, if I make odbc show I see that the DB connection is UP, but if I make realtime load family column value both with extensions family or with sipusers family, I can't find anything in the db. Why it happens? What can I check in my configuration? Someone know if there is a way to test if asterisk make effectively the query to the DB when I make the realtime load command? Please, help me! Maury ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users paste your relevant config files and also an example command (realtime load etc) that you are using. also.. if u can.. turn on logging(DEBUG) in logger.conf, or better still, go change the code n put in ur own debug lines duznt take too long to figure out where u r going wrong. - Ben ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] accountcode and amaflags?
Hello ppl, Can someone explain to me the meaning and use of the variables accountcode and amaflags in sip.conf,etc. Googled, voip-infoed, wikied, etc for it. Couldnt get much of it. I know, they are billing related, but not much beyond that. Any ideas? cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] accountcode and amaflags?
Giovanni, Appreciate your lines mate. But, Ive already read those, all over the net. my qs inline : amaflags : Categorization for CDR records. Choices are default, omit, billing, documentation and choices are defaul, omit, billing, documentation wot r these categories??wot decides these categories? accountcode : string : Users may be associated with an accountcode (billing purpose) hmm.. ive seen in quite a few places, where the pin collected is stored as the accountcode... wot duz that mean? anyway, can you give me an example of wot the association means?am a lil slow.. Cheers, Giovanni 2006/10/19, Benjamin Jacob [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hello ppl, Can someone explain to me the meaning and use of the variables accountcode and amaflags in sip.conf,etc. Googled, voip-infoed, wikied, etc for it. Couldnt get much of it. I know, they are billing related, but not much beyond that. Any ideas? cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] nat auto detect ?
Eric ManxPower Wieling wrote: Benjamin Jacob wrote: Hello ppl, This post is to do with the variables 'nat' or 'canreinvite' for sip entities. Idealy users, wont be static, they could be roaming all over the globe. So, setting someone as behind NAT, and disabling canreinvite, etc., restricts the roaming capabilities of a user. No. Almost all devices work fine with nat=yes, even if they are not behind NAT. ___ hmm.. ok..let me rephrase my subject, it shud be canreinvite auto detect? The issue is to set canreinvite to yes or no. In an ideal world, the server shud detect, if it should have media passing thru itself, or allow a peer-to-peer audio flow. Ofcourz this behaviour should be controllable. So, the question is, wot do I set canreinvite to?If two users, who are behind two different NATs, and some beautiful morning, step out into the internet, and then make calls, it would be wonderful to let the audio flow between each other directly, thereby offloading the traffic off the *. Any chances? cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Locking phones at night...
Conrad Wood wrote: On Wed, 2006-10-18 at 13:24 +0200, Tzafrir Cohen wrote: On Tue, Oct 17, 2006 at 09:42:37PM +0100, Conrad Wood wrote: On Wed, 2006-10-18 at 08:55 +1300, Hadley Rich wrote: On Wednesday 18 October 2006 05:47, Conrad Wood wrote: To do something similar, I created a dialplan extension that - if dialled - creates a file on the server. If dialled again, it removes the file again. Then, in the context of the phone I check for existence of that file and if it exists I play a busy signal and hangup. (Of course, unless the extension to re-enable it is dialled ;) ). Additionally, I ask the user for a password to lock/unlock it. This is a good use for the AstDB Sure is, but files in the filesystem are easier to process from external (non-asterisk) programs. In my case, I have a web interface that locks/unlocks phones too. I find it most convenient to use 'ls' to look up the current status of stuff. asterisk -rx could also be used. Or a phone menu. Problems with a phone menu: how can you tell the status? asterisk -rx requires access to the asterisk console which throws its own bunch of problems with permissions and scalability. I'd then prefer to code it through the manager interface but that seems like a terrible overkill here ;) How would you use a phone menu for that? That sounds interesting. Our users here like doing phonestuff on their phones rather than on websites etc. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users am I missing something important over here?? DB, more specificaly, having ODBCput and ODBCget operations solve all these issues, dont they. read the post abt Stopping putgoing calls after working hours (well.. the subject says so!! ) have your astdb in sql. simple. create extensions to lock/unlock phones or even check status using astdb in sql. very easy to add/view/modify from a webpage too. or... again.. am i missing something over here? - Ben ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Locking phones at night...
Tzafrir Cohen wrote: On Wed, Oct 18, 2006 at 05:26:49PM +0530, Benjamin Jacob wrote: Conrad Wood wrote: On Wed, 2006-10-18 at 13:24 +0200, Tzafrir Cohen wrote: On Tue, Oct 17, 2006 at 09:42:37PM +0100, Conrad Wood wrote: On Wed, 2006-10-18 at 08:55 +1300, Hadley Rich wrote: On Wednesday 18 October 2006 05:47, Conrad Wood wrote: To do something similar, I created a dialplan extension that - if dialled - creates a file on the server. If dialled again, it removes the file again. Then, in the context of the phone I check for existence of that file and if it exists I play a busy signal and hangup. (Of course, unless the extension to re-enable it is dialled ;) ). Additionally, I ask the user for a password to lock/unlock it. This is a good use for the AstDB Sure is, but files in the filesystem are easier to process from external (non-asterisk) programs. In my case, I have a web interface that locks/unlocks phones too. I find it most convenient to use 'ls' to look up the current status of stuff. asterisk -rx could also be used. Or a phone menu. Problems with a phone menu: how can you tell the status? asterisk -rx requires access to the asterisk console which throws its own bunch of problems with permissions and scalability. I'd then prefer to code it through the manager interface but that seems like a terrible overkill here ;) How would you use a phone menu for that? That sounds interesting. Our users here like doing phonestuff on their phones rather than on websites etc. Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users am I missing something important over here?? DB, more specificaly, having ODBCput and ODBCget operations solve all these issues, dont they. read the post abt Stopping putgoing calls after working hours (well.. the subject says so!! ) have your astdb in sql. simple. This means that there is a ODBC lookup per call. well.. i believe it wud be beter than running an external script, thru asterisk for every call. gotta test that :-) u'll be eating up cpu, along with asterisk doing its own work. there was some talk abt local dbs n remote dbs and the performance on some voip-info page for asterisk. Cant seem to find it right now. And if the remote database fails, the PBX fails as well. well. thats where redundancy n HA come into picture... for that sake, even the internal Berkely DB could fail. For the sake of simplicity, it might be preferred to use the internal Asterisk DB. aahh.. i wasnt talking of simplistic setups :-) Is there a simple and safe way to query the astdb database outside of Asterisk? as i said. ODBC ops!! cheerz - Ben ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] nat auto detect ?
Hello ppl, This post is to do with the variables 'nat' or 'canreinvite' for sip entities. Idealy users, wont be static, they could be roaming all over the globe. So, setting someone as behind NAT, and disabling canreinvite, etc., restricts the roaming capabilities of a user. Is there any way for Asterisk to auto detect, if a user is behind NAT, also, if two users are behind the same NAT, help in having a peer-to-peer rtp flow between the two users in the call?? cheerz - Ben ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sending sip style messages in response
Hello ppl, Is it possible to send SIP messages as response to the calling UA on failure, for e.g. if a number is blacklisted I would want to send Forbidden to the caller, not just for user comfort but also for testing purposes? I see only Congestion available which sends Service Unavailable. cheerz - Ben ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending sip style messages in response
Nope Mikael, * always seems to send a Decline, n also plays an unavailable file. I tried another scenario, A calls B, B rejects the call. In the tcpdump I see B sending a Forbidden to *, but * sends a Service Unavailable to A. hmm... not too sure, why this decision was made. Mikael Magnusson wrote: Benjamin Jacob wrote: Hello ppl, Is it possible to send SIP messages as response to the calling UA on failure, for e.g. if a number is blacklisted I would want to send Forbidden to the caller, not just for user comfort but also for testing purposes? I see only Congestion available which sends Service Unavailable. Hangup(CALL_REJECTED) or Hangup(21) should work, I think. Mikael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why is this happening?
On Tuesday 17 October 2006 10:31, Time Bandit wrote: The one that never did a mistake, never did anything so the q is.. will you be doing something a lot?? ;-) ... just kidding mate.. but thats a good line neway. cheerz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
