[asterisk-users] Mailing List Shutdown Reminder
Hello, Just a reminder that on February 1st this mailing list will go into a moderated only state meaning new messages will not be accepted. Conversations should move to the community forums[1] to continue them. Archives will remain available. Cheers, [1] https://community.asterisk.org -- Joshua C. Colp Director of Engineering | Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] aeap wss connection
On Tue, Jan 16, 2024 at 9:56 AM marek wrote: > hi, > > i'm trying asterisk AEAP through Haproxy > > > https://docs.asterisk.org/Asterisk_18_Documentation/API_Documentation/Module_Configuration/res_aeap/?h= > > backend speech-gateway-dev-wss > mode http >option forwardfor >option http-server-close >server speech localhost:9811 > > > topology > > Asterisk - Haproxy - Node.js app - Google STT > > > Asterisk - Node.js works ok > > > tests with curl/wsscat are ok > > but asterisk as wss client doesnt work > Looking at the code it doesn't appear as though it was implemented with support for it from what I can tell. -- Joshua C. Colp Director of Engineering | Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP_HEADER GET_TRANSFERRER_DATA chan_pjsip
On Mon, Jan 8, 2024 at 12:07 PM marek wrote: > hi, > > we are moving our asterisk from chan_sip to chan_pjsip > > we are using SIP_HEADER with GET_TRANSFERRER_DATA for headers from > REFER (asterisk - other pbbx - SIP REFER - asterisk) > > > https://github.com/ca4ti/asterisk/commit/4b58609c331c013845a0a61d946cbbc82092170e > > is it supported in pjsip too? or is there other way? > Looking at the REFER implementation[1] it seems like no. You can submit a feature request here[2]. [1] https://github.com/asterisk/asterisk/blob/20/res/res_pjsip_refer.c [2] https://github.com/asterisk/asterisk-feature-requests -- Joshua C. Colp Director of Engineering | Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mailing List Future
On Wed, Dec 13, 2023 at 8:40 AM Joshua C. Colp wrote: > On Mon, Dec 4, 2023 at 8:54 AM Joshua C. Colp wrote: > >> On Mon, Dec 4, 2023 at 8:52 AM Antony Stone < >> antony.st...@asterisk.open.source.it> wrote: >> >>> On Monday 04 December 2023 at 13:39:51, Joshua C. Colp wrote: >>> >>> > The mailing list will not receive emails from the forums. What I was >>> > referring to is that Discourse does provide the ability to receive >>> emails >>> > for posts or threads you're interested in, and you are able to respond >>> over >>> > email to them as well. >>> >>> I use this forum via its email interface, and I agree that it works. >>> The >>> biggest disadvantage I experience is that although you can _reply_ to a >>> thread >>> via email, you cannot create a new one; you have to use the web forum >>> interface for that. >>> >>> I don't know whether the forum software used here could be modified to >>> allow >>> that - I raised the same point on the FreeSwitch forum and an admin >>> quite >>> happily turned it on. Maybe that could be investigated here? >>> >> >> It actually is turned on for some of the categories, but as it's a hosted >> instance I am limited in the available plugins and modification that can be >> done to make this more clear. We can add documentation on the docs site for >> it, and see if we can do something else. >> > > To follow-up on this, I reached out to Discourse and they gave a > suggestion on how to make it more evident (though not as nice as I would > hope). I'll be experimenting with it in January, that is: making it more > clearer/evident that email exists. > Just a reminder all regarding the time frame on the asterisk-users list, and the move to Discourse[1]. In regards to starting threads using email I have gone through and set up email addresses for the various categories. The hard part is communicating this, and the options Discourse gave weren't exactly the best. For the first attempt I have done the following: 1. Added a menu item at the top for "Starting Threads Over Email" 2. Created a forum post[2] which documents the categories and their email address If there's any other suggestions on it feel free to raise it. Cheers, [1] https://community.asterisk.org/ [2] https://community.asterisk.org/t/starting-threads-over-email/100275 -- Joshua C. Colp Director of Engineering | Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mailing List Future
On Mon, Dec 4, 2023 at 8:54 AM Joshua C. Colp wrote: > On Mon, Dec 4, 2023 at 8:52 AM Antony Stone < > antony.st...@asterisk.open.source.it> wrote: > >> On Monday 04 December 2023 at 13:39:51, Joshua C. Colp wrote: >> >> > The mailing list will not receive emails from the forums. What I was >> > referring to is that Discourse does provide the ability to receive >> emails >> > for posts or threads you're interested in, and you are able to respond >> over >> > email to them as well. >> >> I use this forum via its email interface, and I agree that it works. The >> biggest disadvantage I experience is that although you can _reply_ to a >> thread >> via email, you cannot create a new one; you have to use the web forum >> interface for that. >> >> I don't know whether the forum software used here could be modified to >> allow >> that - I raised the same point on the FreeSwitch forum and an admin quite >> happily turned it on. Maybe that could be investigated here? >> > > It actually is turned on for some of the categories, but as it's a hosted > instance I am limited in the available plugins and modification that can be > done to make this more clear. We can add documentation on the docs site for > it, and see if we can do something else. > To follow-up on this, I reached out to Discourse and they gave a suggestion on how to make it more evident (though not as nice as I would hope). I'll be experimenting with it in January, that is: making it more clearer/evident that email exists. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mailing List Future
On Mon, Dec 4, 2023 at 8:52 AM Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Monday 04 December 2023 at 13:39:51, Joshua C. Colp wrote: > > > The mailing list will not receive emails from the forums. What I was > > referring to is that Discourse does provide the ability to receive emails > > for posts or threads you're interested in, and you are able to respond > over > > email to them as well. > > I use this forum via its email interface, and I agree that it works. The > biggest disadvantage I experience is that although you can _reply_ to a > thread > via email, you cannot create a new one; you have to use the web forum > interface for that. > > I don't know whether the forum software used here could be modified to > allow > that - I raised the same point on the FreeSwitch forum and an admin quite > happily turned it on. Maybe that could be investigated here? > It actually is turned on for some of the categories, but as it's a hosted instance I am limited in the available plugins and modification that can be done to make this more clear. We can add documentation on the docs site for it, and see if we can do something else. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mailing List Future
The mailing list will not receive emails from the forums. What I was referring to is that Discourse does provide the ability to receive emails for posts or threads you're interested in, and you are able to respond over email to them as well. On Mon, Dec 4, 2023 at 8:38 AM John Novack wrote: > > > Frank Vanoni wrote: > > On Mon, 2023-12-04 at 08:00 -0400, Joshua C. Colp wrote: > > > >> To that end, we’ve decided to discontinue the mailing lists effective > >> February 1st, 2024. > > That's a very sad news! :-( > > > Agree. Yet another giant step backward. > Interesting that they will continue to send e-mails when postings to the > (UGH) forum happen though. > > John Novack > > > > -- > Dog is my Co-Pilot > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mailing List Future
Greetings all, Over the past few years, the use of the Asterisk mailing lists has diminished, with far more conversation happening on the Asterisk community forums[1]. The state of email, to ensure reliable delivery, has also gotten more complicated - emails get caught by spam filters, etc.. To continue the mailing lists would require a huge time and resource investment, for minimal use. To that end, we’ve decided to discontinue the mailing lists effective February 1st, 2024. This means the following: 1. Sending and receiving mailing list emails will no longer be possible. 2. The list archives, however, will remain available. We recommend those who have not already done so migrate to the Asterisk Community forums[1]. You can choose to receive emails for posts if you wish, or purely use the web interface. You’re also able to privately message other individuals if you wish. Scoped categories also exist for more specific help. Cheers, [1] https://community.asterisk.org/ -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Finding old patches
On Mon, Nov 20, 2023 at 1:45 PM Dovid Bender wrote: > Hi, > > In the past when I wanted to back port a patch I would go on to the issue > tracker and find a link to the patches that were uploaded ( I think > through gerrit?). I am trying to see what changes were done for > https://issues-archive.asterisk.org/ASTERISK-26109. It seems the code > changes were introduced in 14.4.0-rc1. Is there any "n00b" way of seeing > what patches were created for this specific issue? > The git commit history can be used, the commit message can be grepped, example: ✔ jcolp@kappa:~/development/asterisk/github [21| …2⚑ 3]> git log | grep -B4 "OpenSSL 1.1.0 support" Merge: a0c0b1c9cb 26c8552fff Author: zuul Date: Wed Nov 30 23:26:46 2016 -0600 Merge "OpenSSL 1.1.0 support" -- commit 26c8552fff499419bdf12b663e76ecfc408b3085 Author: Tzafrir Cohen Date: Tue Jun 28 23:26:59 2016 +0200 OpenSSL 1.1.0 support So the commit is "26c8552fff499419bdf12b663e76ecfc408b3085" and you can use git show to display that commit, which includes the changes including in diff format: ✔ jcolp@kappa:~/development/asterisk/github [21| …2⚑ 3]> git show 26c8552fff commit 26c8552fff499419bdf12b663e76ecfc408b3085 Author: Tzafrir Cohen Date: Tue Jun 28 23:26:59 2016 +0200 OpenSSL 1.1.0 support OpenSSL 1.1.0 includes some major changes in the interface. See https://wiki.openssl.org/index.php/1.1_API_Changes . Status: Right now there are still a few deprecation notes with OpenSSL 1.1.0. But it's a start. Changes: * CRYPTO_LOCK is no longer available. Replace it with its value for now. I don't completely understand what it is used for there. * Remove several functions from libasteriskssl that seem to no longer be needed. * Structures have become opaque and are accesses with accessors. * ERR_remove_thread_state() no longer needed. * SSLv2 code now could no longer be used in 1.1. ASTERISK-26109 #close Change-Id: I5e29d477d486ca29b6aae0dc2f5dff960c1cb82b diff --git a/main/iostream.c b/main/iostream.c The same goes for the rest of the associated commits on the linked issue. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Local calls not possible when Internet connection down
On Tue, Nov 7, 2023 at 11:20 AM Marek Greško wrote: > Hello, > > well I do not ask those who only guess, but those who know what is > asterisk expected to do when internet connectivity goes down. I did not had > a chance to make internet not to work yet, since it is needed. But > inspecting dns logs I found out that there started to be resolving for > _sip._tcp and _sip._udp records for the provider's server. So apparently > making hosts record make asterisk happy when everything works, but when > there is a communication problem then it falls back to searching for srv > records. At least it seems to be so for now. Moreover I found out this old > thread: > The expectation is that Asterisk continues to work. That being said there is one case (specifically using realtime with an identify section that references a hostname) that can cause this specific behavior where PJSIP will block. Are you in that scenario? If so you CAN disable SRV records on the identify by setting "srv_lookups" to "no". -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Local calls not possible when Internet connection down
On Mon, Nov 6, 2023 at 10:42 AM Marek Greško wrote: > It looks like all phones get unregistered, but I am not aware of the > cause. Why are get not registered when there is a connectivity between them > and asterisk? > Are the REGISTER requests reaching Asterisk (do they show up in a packet capture, do they show up in "pjsip set logger on")? It needs to be further isolated. How are the phones configured to reach Asterisk? If using a hostname, are they still able to resolve it? -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Local calls not possible when Internet connection down
On Mon, Nov 6, 2023 at 10:06 AM Marek Greško wrote: > Hello, > > I just realized that when my Internet connection goes down and I loose > connectivity to VoIP SIP provider I loose ability to make local calls after > some time. When I restart asterisk, I am able to make local calls for some > time, but it then suddenly stops working again. I am using pjsip stack. > > What could be the cause of this? > There is insufficient information to be able to answer this. Such as, what actually happens when attempts are made? What shows on the console? -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
An issue[1] was already created by aster...@phreaknet.org and they also put a fix up for review and inclusion[2]. [1] https://github.com/asterisk/asterisk/issues/308 [2] https://github.com/asterisk/asterisk/pull/309 On Wed, Sep 13, 2023 at 4:27 PM Jerry Geis wrote: > > I have found that I can add the restart of asterisk (killall -9 asterisk) > to the h extension- BOY is that UGLY. > > chan_console is not a testing device - how can we get this nasty bug fixed > ? > > Jerry > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
On Thu, Sep 7, 2023 at 3:27 PM Jerry Geis wrote: > > I found "console list available" > > === > === - > === Device Name: default > === ---> Default Input Device > === ---> Default Output Device > === - > === > === - > === Device Name: dmix > === ---> Output Device > === - > === > = > > dmix is there and default is there > I tried both - and get the same error > Console device "dmix" not found . etc. > Yes, because that lists the available devices. You have to configure it in console.conf in order to be able to dial it. If you haven't configured a thing named "dmix" in console.conf, then it's not going to work. "console list available" show available devices that you can use in the configuration "console list devices" show what is actually configured -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
On Thu, Sep 7, 2023 at 3:20 PM Joshua C. Colp wrote: > On Thu, Sep 7, 2023 at 3:15 PM Jerry Geis wrote: > >> Joshua >> >> Asterisk 18.14.0 with chan_alsa and Console/dsp works. >> This does not work in 18.18.0 with chan_console enabled. >> I am on Ubuntu 20.04 LTS. >> >> Is there a howto for the new chan_console ? >> > > I'm not aware of one. The module itself has existed since at least > Asterisk 1.8 > > >> how can I get this working again ? >> I am trying to just play audio on pulse audio. >> > > I don't have anything additional to add beyond what I've said and the > config file I've provided. > I can say that with the default configuration file it would be Console/default though, and would use the system default input and output devices according to PortAudio. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
On Thu, Sep 7, 2023 at 3:15 PM Jerry Geis wrote: > Joshua > > Asterisk 18.14.0 with chan_alsa and Console/dsp works. > This does not work in 18.18.0 with chan_console enabled. > I am on Ubuntu 20.04 LTS. > > Is there a howto for the new chan_console ? > I'm not aware of one. The module itself has existed since at least Asterisk 1.8 > how can I get this working again ? > I am trying to just play audio on pulse audio. > I don't have anything additional to add beyond what I've said and the config file I've provided. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 18.18.0 and chan_console
On Thu, Sep 7, 2023 at 3:07 PM Jerry Geis wrote: > > I am trying to get audio to play on Pulse - so just the monitor basically. > > I have tried Console/dsp, Console/dmix, Console/pulse, and a couple others. > > The error is always the same "console_request: Console device 'dmix' not > found. > > What is the correct "Console/" to play on pulse for UBuntu 20.04 LTS ? > I can "aplay /usr/share/sounds/alsa/Front_Center.wav" no problem. > > Thoughts? > It has a configuration file[1] that defines the various devices and their referenced name. If default is in use then I'd expect Console/default [1] https://github.com/asterisk/asterisk/blob/master/configs/samples/console.conf.sample -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 18.18.0 and chan_console
On Wed, Sep 6, 2023 at 12:01 PM Jerry Geis wrote: > >> Just to verify that you did rerun configure after installing the >> libraries? >> >> Doug >> > > Oh that is a good one - I thought I did - but apparently not. menuconfig > now shows "*" > > So is chan_alsa going away ? What is it being replaced with? > The chan_alsa module has been removed in Asterisk 21[1]. The recommended replacement is chan_console. [1] https://docs.asterisk.org/Development/Asterisk-Module-Deprecations/ -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstriCon 2024: February 15th, 2024 - Fort Lauderdale, Florida
Greetings all, The 25th anniversary of Asterisk is upon us! We’ll be celebrating it at AstriCon 2024 held on February 15th, 2024 in Fort Lauderdale, Florida as part of IT Expo. We’d love it if you would join us. You can register here[1], and if you would like to speak you can submit a speaking proposal by October 14th here[2]. I look forward to seeing many of you there! While not AstriCon specific I'm also doing a webinar next week called "The Evolution of Asterisk" which will talk about the history, how Asterisk has changed including philosophy on things, and give a glimpse into future interests. You can register for that here[3]. I can promise I will say words and they might mean things. Cheers, [1] https://www.astricon.net/ [2] https://docs.google.com/forms/d/e/1FAIpQLSeECYs1Bltl0wdfEF57ngGiorTAq79l5Du-Uyn_MuvAut57jw/viewform [3] https://register.gotowebinar.com/register/969603661543016283 -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ICE Candidate collision on dualstack hosts?
On Wed, Aug 23, 2023 at 4:36 AM Benoît Panizzon wrote: > Hi > > I'm attempting to use ICE to be able to present all possible RTP > transports to peers. > > 16.28.0~dfsg-0+deb11u2 (I know it's old, but unfortunately Asterisk was > removed from debian 'stable' and the version in 'sid' is just broken > (opus + voicemail don't work anymore). > > But I ran into an issue when the peer is running rtpengine: > > Asterisk offers: > > a=candidate:H9da13901 1 UDP 2130706431 157.161.57.1 13104 typ host > a=candidate:H1054cffa 1 UDP 2130706431 2001:4060:dead:beef::1 13104 typ > host > a=candidate:He9b56028 1 UDP 2130706431 fe80::5054:ff:fea2:9057 13104 typ > host > a=candidate:H9da13901 2 UDP 2130706430 157.161.57.1 13105 typ host > a=candidate:H1054cffa 2 UDP 2130706430 2001:4060:dead:beef::1 13105 typ > host > a=candidate:He9b56028 2 UDP 2130706430 fe80::5054:ff:fea2:9057 13105 typ > host > > To me this looks like every candidate is duplicated on port +1 > > rtpengine complains: > > [ice] Priority collision between candidate pairs > sKy64vK5pY86kc9w:H9da13901:2 and sKy64vK5pY86kc9w:H9da13901:2 - ICE will > likely fail > > And indeed RTP starts on IPv6 as proposed by H1054cffa but as soon as a > re-invite is processed rtpengine switches to I guess H9da13901 and rtp dies. > > Why is asterisk proposing two ports per ip protocol? Is there a way to > configure this more precisely? > They are not strictly duplicated. They are candidates for different components. One is for RTP, one is for RTCP. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some links on new docs asterisk org not working
On Tue, Aug 22, 2023 at 11:47 AM Dan Cropp wrote: > Not sure where to mention this. Very minor/trivial issue. Just wanted to > let someone know. > The documentation site is stored on Github[1] so issues should be filed there, and people can fix things if they wish by submitting a PR. [1] https://github.com/asterisk/documentation -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 60+ devices in confbridge and dropping audio
On Mon, Aug 21, 2023 at 3:00 PM Jerry Geis wrote: > >> What is a one-way conf? >> >> >> >>* 60+ devices and packets choppy or dropping audio. >> *> >> How have you determined the packets choppy/dropping audio? >> >> >> >>* The CPU is decent at Intel(R) Xeon(R) CPU E3-1240 v5 @ 3.50GHz >> *>>* What else might I tweak to get this working without audio dropping ? >> *>* not much else is running on the server besides - asterisk. >> *> >> Is it virtualized? What timer is being used? >> >> > one way conf is ME - setting up users to be Muted. > [ConfUserMuted] > type=user > quiet=yes > startmuted=yes > announce_only_user=no > announce_user_count_all=no > announce_join_leave=no > > The people at that environment shared that the audio is choppy or drops > out. > > The server is Virtual server. How do you tell what "timer" is used ? > The "timing test" CLI command will state it. Does the VM have guaranteed resources? -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 60+ devices in confbridge and dropping audio
On Mon, Aug 21, 2023 at 11:24 AM Jerry Geis wrote: > I am using asterisk 18.14.0 and chan_sip. > confbridge has dsp_drop_silence=yes > The conf joins all the endpoints in a one-way conf. > What is a one-way conf? > > 60+ devices and packets choppy or dropping audio. > How have you determined the packets choppy/dropping audio? > > The CPU is decent at Intel(R) Xeon(R) CPU E3-1240 v5 @ 3.50GHz > > What else might I tweak to get this working without audio dropping ? > not much else is running on the server besides - asterisk. > Is it virtualized? What timer is being used? -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segmentation fault
On Wed, Aug 16, 2023 at 7:48 PM Federico wrote: > I tested this issue with version 13 and version 18. > > In res_odbc.conf, if I add a second, new data source like > > > > [asterisk] > > enabled=yes > > dsn=asterisk > > sanitysql => select 1 > > isolation => read_committed > > username=root > > ;password= > > pre-connect => yes > > forcecommit => yes > > connect_timeout => 10 > > negative_connection_cache => 0 > > max_connections =>500 > > > > my odbc.ini > > [cdr] > > Description = MySQL ODBC Driver Testing > > Driver = maria > > Socket = /var/run/mysqld/mysqld.sock > > User = root > > Password = > > Database = public > > Option = 3 > > > > > > I get, immediately, segmentation fault. > > With only one, it works fine. > > Is this by design? > There's nothing explicitly written to prevent such a thing within Asterisk itself. There is no backtrace here so nothing to show where the crash actually occurred, be it Asterisk itself or UnixODBC. If UnixODBC we're a somewhat simple user of it, so specific configuration of it or its build may be the source of it. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alternative to Local channel
On Fri, Aug 18, 2023 at 12:38 AM Federico wrote: > It's a great idea but it doesn't work. > Maybe this should be the way that works. > I just did it with stock sample configs aside from adding the variable to globals and it worked perfectly fine. Any modules which contain any referenced dialplan functions need to load before pbx_config. This can explicitly be done using modules.conf -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP Losing knowledge of external_media_address
On Fri, Aug 18, 2023 at 1:09 PM Mark Murawski wrote: > I've seen this happen three times in the wild now. I've been trying to > isolate the source of the issue, but so far it seems like there's not > enough debug output to know why this occurs. > > Long story short: > - Start Asterisk > - PJSIP Handles receiving INVITE from ITSP via WAN (Asterisk is behind > NAT). SIP is handled correctly, Asterisk responds OK with RTP media > address of external_media_address > - After 30 minutes to an hour or sometimes months later after startup, > upon receiving INVITE from ITSP via WAN, Asterisk responds OK with > INTERNAL LAN IP instead of external_media_address > - I've observed this occur after 30 minutes from startup with no > configuration changes that were made or any pjsip reloads done during > this period > > > > > Attached sip sessions and debug log... the only thing I found > interesting was finding a lack of a log item > We SHOULD be seeing: > DEBUG[X] res_pjsip_session.c: (null session): Setting external media > address to 152.X.Y.Z > This message is clearly lacking from the debug session where the > incorrect media address is sent. But there's not enough detail in the > debugs to see why this decision was not made to use external_media_address > Can't you just extend the debug and add further logging to understand the choices being made and why? > > By default we use nat settings for all our endpoints, but obviously it's > not required here for an ITSP that has trustworthy media ports in the > SDP. Maybe a bandaid is turning off rewrite_contact for this endpoint? > Going to try that as soon as possible. > I believe I've stated this once or twice when you've brought this issue up on IRC but rewrite_contact has no influence or impact on this. It rewrites incoming Contact headers to the source IP address and port of the SIP message. You can turn it on if you wish, but it is unlikely to do anything. > > Why is external_media_address not being used all of a sudden? Has > anyone else seen this... is this a bug?-- > With the limited insight into things it could be a bug. I haven't seen any other reports, and haven't received any reports from other Sangoma products. Is this with a mainline Asterisk, or is it your patched version of Asterisk? It should be confirmed on normal Asterisk. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Encountered a crash, what is best way to tell if it has been fixed or now
On Wed, Aug 9, 2023 at 4:11 PM Dan Cropp wrote: > I was able to put the crash through the gdb on the original VM that > encountered the problem. > > (Not sure why the VM I initially tried to analyze the crash dump on didn’t > do this correctly, but not concerned about it now). > > > > It’s providing additional details. > That is closer. There's still some optimized elements, mostly in PJSIP from what I can tell, but it would be usable probably. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Encountered a crash, what is best way to tell if it has been fixed or now
On Wed, Aug 9, 2023 at 3:20 PM Dan Cropp wrote: > I have a customer who just encountered a crash while running Asterisk > 18.17.1 version. > > > > I’m trying to adapt to the changes so not sure where best to look or how > to possibly report this. > > > > I started by going through > https://github.com/asterisk/asterisk/compare/18.17.1...18.19.0 to see if > any of the changes seemed to apply to code reported by the backtrace. > > > > Entirely possible I missed something, but I didn’t notice anything that > applies. > > > > I do see a commit was done today to the res_pjsip_nat.c file, but not sure > if that would apply to the issue. > > > > Any suggestions for where I should look or ask? > That is how you generally look, by seeing the commits between the two versions, analyzing, and seeing if anything is relevant. Issues themselves are reported on Github. I can say already though that the backtrace is incomplete and doesn't show the full story of what happened, it may be optimized or something. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Subscribing to events on AMI login
On Tue, Aug 8, 2023 at 1:29 PM TTT wrote: > Ok – so if I forgot to add “security” to the read= line in manager.conf > for this user, will that cause the user to be unable to subscribe to the > “security” events upon login? (in other words, although subscribed at > login, no security events will be shown to this user) > Yes. The user needs to have the permissions configured on them. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Subscribing to events on AMI login
On Tue, Aug 8, 2023 at 12:44 PM TTT wrote: > I'm looking at an old app I wrote that upon AMI login would subscribe to > events as follows: > > > > Action: Login > > ActionID: myid > > Username: myun > > Secret: mypw > > Events: call, system, security > > > > I noticed this old code isn't working, and I *think* that the events > parameter of login has been deprecated; I don't see it referenced in: > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_Login > > > > I’m using Asterisk 20, so Is the events parameter still valid? I don't > seem to receive any events other than the "FullyBooted" event after login. > If not valid, how should I subscribe to events programmatically? > The parameter appears to be valid, just implemented in such a way that it likely got missed when writing the documentation. As for not working, you'd need to provide the manager.conf configuration as well. There is also the Events AMI action[1] for changing it after login. [1] https://docs.asterisk.org/Asterisk_20_Documentation/API_Documentation/AMI_Actions/Events/ -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can ShanSpy be used on Local Channels?
On Tue, Jul 25, 2023 at 6:06 PM Carlos Chavez wrote: > Does anyone know if Chanspy can be used with local channels? Since > agents on queues need to use local channels like Local/@from-queue/n > to make sure that all of their registered extensions ring we are now > having a problem trying to use ChanSpy to listen to calls. Since we do > not know if the agent is on their desk phone or a softphone (which use > different identifiers) we cannot set a common rule like > ChanSpy(PJSIP/). Queuemetrics registers agent extensions as: > Local/@from-queue but if I try to listen on that channel I get no > audio. Am I missing a parameter or is chanspy simply unable to use > Local channels? > ChanSpy works on any channel. You'd need to provide more information and detail to determine what is wrong. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI versions
On Tue, Jul 11, 2023 at 3:40 PM Joshua C. Colp wrote: > On Tue, Jul 11, 2023 at 3:38 PM TTT wrote: > >> That answers part two…but is there any mapping of AMI version to Asterisk >> versions? >> > > No, there is not. > I can say that Asterisk 13 is 2.x.x though because I just looked, so you can use the version policy to determine what each release is subsequently. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI versions
On Tue, Jul 11, 2023 at 3:38 PM TTT wrote: > That answers part two…but is there any mapping of AMI version to Asterisk > versions? > No, there is not. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Release 20.3.1
On Fri, Jul 7, 2023 at 6:40 PM Jean-Denis Girard wrote: > There seems to be a problem with the tar.gz archive on github. It's > correct on downloads.asterisk.org. Can you be more specific? They are identical and the same tarball. I just downloaded both from each place and confirmed that, and confirmed they both extract fine. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
On Thu, Jul 6, 2023 at 2:22 PM Michael Ulitskiy wrote: > Oh, that's great. It wasn't clear from that page, at least not for me. :-( > > Having it clearly stated on the document would save me (and probably > others) lots of time. > The wiki is read-only now and documentation has moved to https://docs.asterisk.org/, I have updated the page[1] and it will deploy in a few minutes. > Thanks for clarifying it. Any idea on the timeframe of implementation? > There is no timeframe on such a thing. [1] https://docs.asterisk.org/20/Development/Roadmap/Asterisk-18-Projects/Advanced-Codec-Negotiation-ACN/?h=advanced+codec -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
On Thu, Jul 6, 2023 at 1:43 PM Michael Ulitskiy wrote: > Hello, > > After I have re-read the "PJSIP Advanced Codec negotiation" document, it > occurred to me that the desired behavior should actually happen > automatically, just due to the codec negotiation logic, but it looks like > asterisk doesn't actually follow the described logic which is likely a bug. > > That functionality is not implemented as of this time. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getvar of CHANNEL not working for a couple of items
On Wed, Jul 5, 2023 at 12:50 PM TTT wrote: > Channel A: "1688509741.112" , name: "PJSIP/111-0064" , is > originator: Y , call-Id: "u.l6kcou25ca...@mydomain.com" , local_uri: "< > sip:2...@mydomain.com;user=phone>" , local_tag: > "1734d973-c4da-4ae8-a37d-5f7065f1fe54" , local_addr: "172.31.253.4:5060" > , remote_uri: "\\\"TestPhone x111\\\" " , > remote_tag: "yinue4v5ufa4" , remote_addr: "172.31.253.20:5060" > > > > > > Channel B: "1688509741.113" , name: "PJSIP/222-0065" , is > originator: N , call-Id: "1f104544-fc1a-4414-ba74-68c526e294de" , > local_uri: "\\\"TestPhone\\\" " , local_tag: > "ac5eeb59-f559-4bb7-a3c2-170ca7f05f8b" , local_addr: "" , remote_uri: "< > sip:222@172.31.253.20;line=46922>" , remote_tag: "klwqxe1fvt5wk" , > remote_addr: "" > > > > And here's what seems strange: > > Channel A's local_uri looks like Channel B's uri > > Channel A's remote_uri looks like channel A's uri > > Channel B's local_uri looks like channel A's uri > > Channel B's remote_uri looks like channel B;s uri > > > > These aren't strange. They look alike because of callerid and target > dialed information. They are still independent call legs. > > > > > > I’m having trouble understanding your explanation (googling just led me to > generic callerid and target info). I thought a phone’s local_uri would be > how to reach that phone (not the other party), and vice versa for the > remote_uri. If the above URI info is correct then I must misunderstand > their meaning. Could you provide more explanation on how to interpret them > (why they seems reversed to me), or a link? > > > > I assumed the remote & local URI where equivalent to the to & from lines > (respectively) in the invite… > They are the From and To header, but what remote_uri and local_uri refers to changes depending on the direction of the SIP dialog. Received call: From = remote_uri, To = local_uri Sent call: From = local_uri, To = remote_uri The contents of each depend on callerid information, settings, the Contact of the target when doing an outgoing call, what the remote endpoint chose for To URI on a received call. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getvar of CHANNEL not working for a couple of items
On Tue, Jul 4, 2023 at 7:52 PM TTT wrote: > Building on my last message, I am trying to get CHANNEL data using getvar > (through the AMI). And although I'm getting responses, some values > returned seem illogical. For example, phone 111 calls phone 222 via the > PBX. Here's the data I get back > > > > > > Channel A: "1688509741.112" , name: "PJSIP/111-0064" , is > originator: Y , call-Id: "u.l6kcou25ca...@mydomain.com" , local_uri: "< > sip:2...@mydomain.com;user=phone>" , local_tag: > "1734d973-c4da-4ae8-a37d-5f7065f1fe54" , local_addr: "172.31.253.4:5060" > , remote_uri: "\\\"TestPhone x111\\\" " , > remote_tag: "yinue4v5ufa4" , remote_addr: "172.31.253.20:5060" > > > > > > Channel B: "1688509741.113" , name: "PJSIP/222-0065" , is > originator: N , call-Id: "1f104544-fc1a-4414-ba74-68c526e294de" , > local_uri: "\\\"TestPhone\\\" " , local_tag: > "ac5eeb59-f559-4bb7-a3c2-170ca7f05f8b" , local_addr: "" , remote_uri: "< > sip:222@172.31.253.20;line=46922>" , remote_tag: "klwqxe1fvt5wk" , > remote_addr: "" > > > > And here's what seems strange: > > Channel A's local_uri looks like Channel B's uri > > Channel A's remote_uri looks like channel A's uri > > Channel B's local_uri looks like channel A's uri > > Channel B's remote_uri looks like channel B;s uri > These aren't strange. They look alike because of callerid and target dialed information. They are still independent call legs. > Channel B's local_addr is blank > > Channel B's remote_addr is blank > I don't know why they're blank. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get channel variables via ARI/AMI
On Mon, Jul 3, 2023 at 12:28 PM TTT wrote: > The uppercase command made a difference. I now get a call-id as show > below. However, does the call-id look valid? The @0.0.0.0 seems strange. > > > > action: Getvar > > actionid: act1 > > channel: PJSIP/Twilio-NA-W-3-In-0028 > > Variable: CHANNEL(pjsip,call-id) > > > > > > Response: Success > > ActionID: act1 > > Variable: CHANNEL(pjsip,call-id) > > Value: 4decf884e3ae74595906283a74f7154e@0.0.0.0 > Call-ID within a SIP dialog is completely opaque. It is what it is. > > > > > As well, can I request many pieces of data at once? The syntax on this > page ( > https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL) > seems to suggest you pass a single parameter, “item”, yet passing just > call-id did not work. I had to pass “pjsip,call-id”. Is the first > parameter a category and the second the detailed item? What if I want to > retrieve multiple items (or all “pjsip” items)? > You can not request multiple at once. Some things require an initial specifier such as pjsip, some things don't. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get channel variables via ARI/AMI
On Sun, Jul 2, 2023 at 4:39 PM TTT wrote: > >> You use the AMI action Getvar[1] which allows channel variables and > dialplan functions. > > >> [1] > https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_Getvar > > > > > I actually tried that, and although I get “success” I never get useful > data. For example: > > > > action: Getvar > > actionid: act1 > > channel: PJSIP/Twilio-NA-W-2-In-0025 > > Variable: channel(pjsip,call-id) > > > > Response: Success > > ActionID: act1 > > Variable: channel(pjsip,call-id) > > Value: > CHANNEL(pjsip,call-id) Dialplan function names are case sensitive. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get channel variables via ARI/AMI
On Sun, Jul 2, 2023 at 4:18 PM TTT wrote: > >> There are SOME protocol level things accessible using CHANNEL[1] but > that's it. > > >> [1] > https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL > > > > > > I am trying to use the CHANNEL function listed above from the AMI. Since > it is not an AMI “action”, but rather a dialplan “function”, I’m trying to > figure out how to call this from the AMI. Using a telnet session to the > AMI I’ve tried variations of: > You use the AMI action Getvar[1] which allows channel variables and dialplan functions. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_Getvar -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with PJSIP contacts being "unavailable"
On Tue, Jun 27, 2023 at 8:22 AM wrote: > Trace from an "unavailable" ATA (not working correctly): > https://paste.interlinked.us/iz07sapwrb.txt > > Trace from an "available" ATA (working correctly): > https://paste.interlinked.us/ocutyjslmg.txt The "unavailable" ATA no longer has a working TLS connection to Asterisk, resulting in OPTIONS failing, and going unreachable, and eventually the Contact going away. Actually examining the TLS side would be needed. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get channel variables via ARI/AMI
On Mon, Jun 26, 2023 at 4:35 PM TTT wrote: > I think that’s getting me close. I’m trying to get (or recreate) the FROM > and TO lines of the header, from a system running PJSIP. I think if I use > CHANNEL to get local_uri and local_tag I can recreate a FROM line like: > > *FROM=;tag=TAG* > > > > And if I use CHANNEL to get remote_uri and remote_tag I can recreate a > FROM line like: > > *TO=;tag=TAG* > > > > Would it be correct to assume that with this info (and ip:port info) I > should be able to send a UDP SIP message from the PBX to the UA which > appears to be part of the current call dialog? I realize this is an odd > thing to do, but I’m just interested in technical feasibility at this > point. Before I try to code this I want to ensure I’m not missing > something stupid. > Probably not. Sequence number also matters. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get channel variables via ARI/AMI
On Mon, Jun 26, 2023 at 4:04 PM TTT wrote: > It looks like if I call Getvar and pass PJSIP_HEADERS() I can get the > entire SIP header for a channel. I also read (on stackoverflow) that the > PJSIP_HEADER function will only return the headers from the INVITE of the > *inbound* channel. > > > > If that’s correct, how would I get the headers from the outbound channel > (second leg of the bridged call) INVITE ? Or will PJSIP_HEADERS() in fact > return the header from either inbound out outbound legs? > The answer is, you can't. There are SOME protocol level things accessible using CHANNEL[1] but that's it. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get channel variables via ARI/AMI
On Mon, Jun 26, 2023 at 10:57 AM TTT wrote: > I am connecting to the ARI with subscribe all, so I can see channels being > created. I now want to extract a variety of header variables (at the > moment the from and to tag). I tried to read them from the ARI but > Asterisk refuses since the channel is not in a stasis app. > > > > Is there a way to read these from either the ARI or AMI ? I’m trying not > to modify the dialplan. > ARI, No. AMI, Yes[1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_Getvar -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why is WebRTC treated differently from regular SIP in Asterisk
On Fri, Jun 23, 2023 at 11:38 PM TTT wrote: > I’m learning about WebRTC clients, and am wondering why Asterisk treats > them differently from any other SIP client. > > > > The media (RTP) should be no different, so the only difference should be > on the signaling side. I noticed that the Asterisk wiki mentions the need > for res_pjsip_transport_websocket, so does that mean Asterisk requires > the signaling to occur over a websocket? > > > > If I used a SIPJS fork which places the signaling over UDP (eg > https://github.com/cwysong85/sipjs-udp) will it just be a regular SIP > client and I shouldn’t have to configure anything special in Asterisk, just > regular PJSIP. > The signaling can go over whatever transport (UDP, Websocket, TCP, TLS). Websockets are commonly used because as I stated in my other response it is what the browser provides. From a media level WebRTC itself is different because it uses additional standards than a regular SIP client. It does ICE, STUN, TURN, DTLS-SRTP (which makes the SDP incompatible with non DTLS-SRTP SDP), and others for media streams, packet loss, and more. Could a normal SIP client use those? Yes. Do they? Usually no. All of this isn't driven by Asterisk, but WebRTC. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WebRTC signaling
On Fri, Jun 23, 2023 at 11:19 PM TTT wrote: > I’m looking at using Asterisk 20 with WebRTC clients (sipjs). I know the > media runs over TCP, but what about the signaling? > > Media doesn't generally go over TCP, it goes over UDP. > > > I read something about signaling over UDP was proposed as part of a webrtc > standard, but can’t find if that was ever ratified or if Asterisk can even > use UDP for the signaling instead of TCP for the signaling. > > > > Does encryption of the signaling (SIPS) change anything? > WebRTC doesn't define signaling. SIP is an option, and the browser provides websockets for its transport. It's all in what the browser supports. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE
On Wed, Jun 21, 2023 at 4:07 PM TTT wrote: > Something perhaps noteworth, since this is a multihomed system I bound the > transport to 172.31.253.4:5060 > > I don't *think* that would cause Asterisk to use that IP in the FROM...at > least it shouldn't. > > Copy/paste from FreePBX forum: It doesn’t touch the From header because it doesn’t matter for normal use. There is a “from_domain” option which can be used to explicitly set the domain portion of the From header. It’s unlikely to be your problem, unless Twilio requires a specific From domain. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple phones on same PJSIP account
On Wed, Jun 21, 2023 at 12:53 PM TTT wrote:
> Ok I've got multiple phone sets registered with the same extension/secret.
>
> However, this causes a strange problem. If I have 3 phone sets registered
> on extension 123, and I then call extension 123 (from extension 456), only
> a SINGLE phone set will ring.
>
> Is this by design or a bug? Does only the most recently registered phone
> set ring when I call the extension? Seems odd...is there a way to change
> it so ALL phones on the same extension will ring? (I'm using SNOM +
> PANASONIC + Aastra phones)
>
For dialing all contacts you have to use the PJSIP_DIAL_CONTACTS dialplan
function[1] which returns a string you pass to Dial(). For example:
Dial(${PJSIP_DIAL_CONTACTS(alice)})
To dial all registered contacts for "alice".
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_PJSIP_DIAL_CONTACTS
--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] Expanding my answering-machine system
On Sun, Jun 18, 2023 at 9:28 AM Steve Matzura wrote: > Sorry, Joshua, I don't understand. One's a filename, one's an extension > number. How are they the same? In other words, why would > 'enter-ext-of-person' be considered a filename? I would think > 'enter-ext-of-person' would be an extension number. > No. One is not an extension number. One is a Background instructed to play a sound file with the filename "enter-ext-of-person". It is not a placeholder where you replace it with an extension number. It's meant to be used as-is, and play back a sound file named "enter-ext-of-person". -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get SIP Call-ID from ARI
On Sat, Jun 17, 2023 at 8:41 PM TTT wrote:
> I tried
>
> GET /channels/{channelid}/variable?variable=CHANNEL(pjsip,call-id)
>
>
>
> But it responds with
>
> "message": "Channel not in Stasis application"
>
>
>
> Since I want to get the call-id for a channel not in stasis I guess that
> won’t work. Similarly, I can’t force the channel through my own code in
> the dialplan, so the PJSIP_HEADER function won’t work. So it looks like
> I’ll have to upgrade my Asterisk test system to get the Call-ID from the
> ARI event. It looks like it was added in Ast 16.
>
>
>
> Out of curiosity, I see that call-id is returned in the “protocol_id”
> field of channel data structure. However, since all channels in the same
> call must have the same Call-ID, how can this data be associated with a
> channel? Wouldn’t it have to be associated with a bridge? The Call-ID
> should not be available until two legs are bridged (I think).
>
All channels in a call do not have the same Call-ID. Each channel has its
own SIP Call-ID (if it is a PJSIP channel) as they are individual call legs
and individual SIP dialogs.
--
Joshua C. Colp
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Sangoma Technologies
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Re: [asterisk-users] Expanding my answering-machine system
On Sat, Jun 17, 2023 at 7:48 PM Steve Matzura wrote: > OK, this is how I thought it's supposed to work. It just confounded me why > the book would say the Playback() and Background() syntax were the same, > then in the very next paragraph give an example that belied that claim. > The syntax is the same. They both take a filename. The example gave a filename of "enter-ext-of-person". You could pass that to Playback, though you would be unable to enter an extension. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get SIP Call-ID from ARI
On Sat, Jun 17, 2023 at 2:55 PM TTT wrote:
> Based on postings it should be possible to get the SIP Call-ID header
> value from the ARI. At what point is this value available ? As well, how
> do I retrieve that value – something like
>
>
>
> GET /channels/{channelId}/pjsip_header?key=Call-Id
>
>
>
> But that doesn’t work.
>
'pjsip_header' is not a valid route. All possible routes are documented on
the wiki, if it's not there then it doesn't exist.
Instead you would use variable[1] to execute the PJSIP_HEADER dialplan
function[2] or a better way would be the CHANNEL dialplan function[3] such
as:
GET /channels/{channelid}/variable?variable=CHANNEL(pjsip,call-id)
Though I haven't tested that.
Newer versions also include the protocol identifier (Call-ID) in the
channel ARI structure[4] which would be in events, or explicitly
retrieved[5].
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-getChannelVar
[2]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_PJSIP_HEADER
[3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL
[4]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+REST+Data+Models#Asterisk20RESTDataModels-Channel
[5]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-get
--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] Add user to conference via ReST/ARI
On Thu, Jun 15, 2023 at 6:08 PM TTT wrote:
> I’m trying to join a user (at SIP/99) into a conference via REST/ARI. I
> want the PBX to call the user, and then join him into an existing
> conference.
>
>
>
> I have created a conference in FreePBX with number 1234, and name "conf".
> Conceptually the steps I have so far:
>
>
>
> 1. Call Application_Dial(SIP/99) (via REST)
>
> 2. Wait for user to answer (via ARI)
>
> 3. Add the channel to a bridge (via REST)
>
>
>
> I'm getting stuck on step #3. Should I call
> Application_BridgeAdd(channel), where channel is provided via ARI event?
> Or do I use Application_ConfBridge(1234)?
>
>
>
> I'm not sure with the latter option if "conference" parameter is the
> conference number (1234), or name ("conf"), or some other value.
>
You can't. The channel has to be sent into the dialplan using continue, and
then invoke the ConfBridge dialplan application.
--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
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Re: [asterisk-users] Problem with pjsip
On Thu, Jun 8, 2023 at 9:41 AM Yves wrote: > Hello everyone. > I allow myself to submit a problem that I can not solve with my VOIP > provider Orange in France > > [2023-06-08 13:19:03] ERROR[185091]: > res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error > configuring endpoint 'Biv_Sortie' - 'from_user' field contains invalid > character '@' > [2023-06-08 13:19:03] ERROR[185091]: config_options.c:798 > aco_process_var: Error parsing from_user=75b55btqu...@orange-obs.fr at > line 0 of >== chan_pjsip.so => (PJSIP Channel Driver) > > 1) Error with "@" character which constitutes URI and authuser see > excerpt from pjsip.conf. > > [transport-udp] > type = transport > protocol=udp > bind=0.0.0.0:5060 > local_net=172.16.1.0/255.255.255.0 > > [reg_orange-obs.fr] > type = registration > retry_interval = 120 > max_retries = 10 > expiration = 120 > transport = transport-udp > outbound_auth = auth_reg_orange-obs.fr > client_uri = sip:+3313445x...@orange-obs.fr > server_uri = sip:orange-obs.fr > > [auth_reg_orange-obs.fr] > type=auth > password=3314C9BA9688C2AA > username = 75b55btqu...@orange-obs.fr > > [Biv_Sortie] > type = aor > contact = sip:75b55btqu...@orange-obs.fr@orange-obs.fr > default_expiration = 3600 > > [Biv_Sortie] > type = identify > endpoint = Biv_Sortie > match = orange-obs.fr > > [Biv_Sortie] > type=auth > username = Biv_Sortie > password=3314C9BA9688C2AA > > [Biv_Sortie] > type=endpoint > context = Isdn_Inbound > dtmf_mode=rfc4733 > disallow=all > allow = g722, alaw, g729 > direct_media=no > trust_id_inbound = yes > send_rpid=yes > from_user = 75b55btqu...@orange-obs.fr > from_domain = orange-obs.fr > language = en > allow_subscribe = yes > auth = Biv_Exit > outbound_auth = Biv_Sortie > aors = Biv_Sortie > > Question how can I solve this character problem "@"? > The from_user is the username. It can't contain "@" or the domain. You've already set from_domain, so just set from_user to the username. > > 2) resolution of the orange-obs.fr DNS. I am attaching an extract from > the documentation that Orange issued in 2015 > > SIP/Internet is described in RFC3261 and following. THE > SIP/IMS is described by 3GPP standards. It's not the same > SIP. > In the Internet world, VoIP machines route > SIP messages to the IP addresses of the FQDNs of the SIP URIs > (VoIP domain). In the 3GPP world, SIP messages are > routed to an I/P-CSCF (depending on whether we are in interco or in > IPBX) which has a different FQDN from the VoIP domain. > > BIV SIP > > – P-CSCF FQDN: pcscfgm.orange-obs.fr, resolved by DNS > voice > – VoIP domain: orange-obs.fr, not resolved by voice DNS. ex : > INVITE sip:0142277...@orange-obs.fr SIP/2.0 > 2 > The VoIP/Internet machine will not be able to determine the address > recipient of SIP messages. > > run the command “nslookup pcscfgm.orange-obs.fr” and > note the returned IP address 217.167.210.X > – add this address in the /etc/hosts file of the PBX: > 217.167.210.X pcscfgm.orange-obs.fr orange-obs.fr You don't need to do /etc/hosts. Set an outbound_proxy on the endpoint and registration like so: outbound_proxy=sip:cscfgm.orange-obs.fr\;lr;\hide This will cause the SIP requests to get sent to "cscfgm.orange-obs.fr" but that won't appear in the SIP signaling. You'll probably have other issues that will require configuration changing, since providers using IMS infrastructure for general SIP always causes that. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Listen to ARI events
On Wed, Jun 7, 2023 at 3:26 PM TTT wrote: > I’ve been looking through the docs (near > https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Applications+REST+API) > and am searching for a list of events I can subscribe to. Is this list > published? Or can I query the ARI for a list of available events? > You don't subscribe to events, you subscribe to event sources[1] but if subscribeAll is passed, you are subscribed to everything. The events are on the wiki[2] with events having base type Event. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Applications+REST+API#Asterisk20ApplicationsRESTAPI-subscribe [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+REST+Data+Models -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Listen to ARI events
On Wed, Jun 7, 2023 at 12:04 PM TTT wrote: > Ok that worked. > > > > Since I have not declared a statis app called “test”, does that mean any > non-existent app name on the URL will subscribe to all system events? (Or > is test a built-in app name) > Applications are not declared or configured anywhere. The act of connecting a websocket with an app name creates them. And no, you have to pass subscribeAll=yes to have the websocket subscribed to all events. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Listen to ARI events
On Wed, Jun 7, 2023 at 10:46 AM TTT wrote:
> I’ve reread the documentation a few times, and what isn’t clear is whether
> I need an app=X parameter in the url. In other words, can I only get
> events for a single named statis app? Or can I get events for the entire
> Asterisk server?
>
>
>
> The command below (without app= parameter) results in no events being
> shown, but no error either.
>
You must specify an app as well. If you don't, it should reply with a 400.
If it's not... then are you connecting to Asterisk? What does the console
say? For example I did the following:
wscat --connect
"ws://kappa:8088/ari/events?api_key=asterisk:asterisk=yes=test"
Which connected successfully and then I did a call which resulted in:
{"type":"ChannelCreated","timestamp":"2023-06-07T10:54:56.295-0300","channel":{"id":"1686146096.1","name":"PJSIP/mytrunk_endpoint-","state":"Down","protocol_id":"","caller":{"name":"Electron","number":"1009"},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"from-external","exten":"s","priority":1,"app_name":"","app_data":""},"creationtime":"2023-06-07T10:54:56.295-0300","language":"en"},"asterisk_id":"0c:c4:7a:ba:b3:5a","application":"test"}
<
{"type":"ChannelDialplan","timestamp":"2023-06-07T10:54:56.295-0300","dialplan_app":"AppDial2","dialplan_app_data":"(Outgoing
Line)","channel":{"id":"1686146096.1","name":"PJSIP/mytrunk_endpoint-","state":"Down","protocol_id":"d89ddae0-6568-4ab4-995b-8feb39366a58","caller":{"name":"Electron","number":"1009"},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"from-external","exten":"s","priority":1,"app_name":"AppDial2","app_data":"(Outgoing
Line)"},"creationtime":"2023-06-07T10:54:56.295-0300","language":"en"},"asterisk_id":"0c:c4:7a:ba:b3:5a","application":"test"}
<
{"type":"Dial","timestamp":"2023-06-07T10:54:56.295-0300","dialstatus":"","forward":"","dialstring":"mytrunk_endpoint","peer":{"id":"1686146096.1","name":"PJSIP/mytrunk_endpoint-","state":"Down","protocol_id":"d89ddae0-6568-4ab4-995b-8feb39366a58","caller":{"name":"Electron","number":"1009"},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"from-external","exten":"s","priority":1,"app_name":"AppDial2","app_data":"(Outgoing
Line)"},"creationtime":"2023-06-07T10:54:56.295-0300","language":"en"},"asterisk_id":"0c:c4:7a:ba:b3:5a","application":"test"}
<
{"cause":34,"type":"ChannelHangupRequest","timestamp":"2023-06-07T10:54:56.296-0300","channel":{"id":"1686146096.1","name":"PJSIP/mytrunk_endpoint-","state":"Down","protocol_id":"d89ddae0-6568-4ab4-995b-8feb39366a58","caller":{"name":"Electron","number":"1009"},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"from-external","exten":"s","priority":1,"app_name":"AppDial2","app_data":"(Outgoing
Line)"},"creationtime":"2023-06-07T10:54:56.295-0300","language":"en"},"asterisk_id":"0c:c4:7a:ba:b3:5a","application":"test"}
<
{"type":"Dial","timestamp":"2023-06-07T10:54:56.296-0300","dialstatus":"CONGESTION","forward":"","dialstring":"mytrunk_endpoint","peer":{"id":"1686146096.1","name":"PJSIP/mytrunk_endpoint-","state":"Down","protocol_id":"d89ddae0-6568-4ab4-995b-8feb39366a58","caller":{"name":"Electron","number":"1009"},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"from-external","exten":"s","priority":1,"app_name":"AppDial2","app_data":"(Outgoing
Line)"},"creationtime":"2023-06-07T10:54:56.295-0300","language":"en"},"asterisk_id":"0c:c4:7a:ba:b3:5a","application":"test"}
<
{"type":"ChannelDestroyed","timestamp":"2023-06-07T10:54:56.296-0300","cause":34,"cause_txt":"Circuit/channel
congestion","channel":{"id":"1686146096.1","name":"PJSIP/mytrunk_endpoint-","state":"Down","protocol_id":"d89ddae0-6568-4ab4-995b-8feb39366a58","caller":{"name":"Electron","number":"1009"},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"from-external","exten":"s","priority":1,"app_name":"AppDial2","app_data":"(Outgoing
Line)"},"creationtime":"2023-06-07T10:54:56.295-0300","language":"en"},"asterisk_id":"0c:c4:7a:ba:b3:5a","application":"test"}
--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
--
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/
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Re: [asterisk-users] Listen to ARI events
On Tue, Jun 6, 2023 at 6:04 PM TTT wrote: > I have the ARI enabled on my Asterisk test box, and want to listen to all > events. I can’t find the syntax to do that. Can I only listen to events > related to a stasis app? > > > > I was hoping that a simple wscat command like this would show me all > events: > > > > wscat -c "ws://localhost:8088/ari/events?api_key=asterisk:asterisk " > This does not listen to all events by default. If you want to listen to everything you can pass subscribeAll=yes[1] like so: ws://localhost:8088/ari/events?api_key=asterisk:asterisk=yes [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Events+REST+API#Asterisk20EventsRESTAPI-userEvent -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SAY_DTMF_INTERRUPT not working
On Tue, May 16, 2023 at 11:01 PM Dovid Bender wrote: > Hi, > > I am trying to use SAY_DTMF_INTERRUPT with Asterisk 20.0.1. I see that I > asked about it here https://www.spinics.net/lists/asterisk/msg174142.html > and Sean was nice enough to create a patch. I am trying it on 20.0.1 by > doing in the dial plan: > CHANNEL(SAY_DTMF_INTERRUPT)=on > and I get back: > Unknown or unavailable item requested: 'SAY_DTMF_INTERRUPT' > When I do: "core show function channel" I don't see it as an option. What > am I doing wrong? > It's a normal channel variable, accessible using Set(SAY_DTMF_INTERRUPT=on) It is not a field on the CHANNEL dialplan function. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opus: No translation path after upgrade ubuntu focal => jammy
On Fri, May 5, 2023 at 6:23 AM Benoît Panizzon wrote: > Hey! > > I just upgraded our machines from Ubuntu focal to jammy. > > A separate package asterisk-opus does not exist any more. > > https://launchpad.net/ubuntu/+source/asterisk-opus/+changelog > > It looks like this is now included in the default packages. > > Required modules are loaded: > > *CLI> module show like opus > Module Description > Use Count Status Support Level > format_ogg_opus_open_source.so OGG/Opus audio 0 > Running core > res_format_attr_opus.soOpus Format Attribute Module 1 > Running core > > *CLI> module show like resample > Module Description > Use Count Status Support Level > codec_resample.so SLIN Resampling Codec0 > Running core > > Core show codecs shows: > > 31 audio opus opus (Opus Codec) > > *CLI> core show translation paths opus > > Shows no translation path to/from any other codec. > > What could I be missing? > There is no codec_opus module loaded, thus no transcoding of it. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Broken link in LICENSE file
On Mon, May 1, 2023 at 12:30 PM John Runyon wrote: > https://github.com/asterisk/asterisk/blob/master/LICENSE#L48 broken > > (PS I hope I never find a bug to report, because I don't use Github... > embrace, extend, extinguish is still alive and well) > I have created a GitHub bug report for it[1] and put up a PR to update it to the correct link, which is https://www.sangoma.com/wp-content/uploads/Sangoma-Trademark-Policy-1.pdf [1] https://github.com/asterisk/asterisk/issues/43 -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk translates 200 OK + SDP into 488 not acceptable here after both side agreed on codec.
On Fri, Apr 28, 2023 at 10:43 AM Benoît Panizzon wrote: > Hi List > > Asterisk 16.28.0 in use. > > PJSIP in use > Two endpoints > Both using IPv6 > > One Endpoint on UDP, the other via TLS. > > Both with: > > t38_udptl=yes > ;fax_detect=yes > ;fax_detect_timeout=30 > rtp_ipv6=yes > > Both sides are T.38 capable and detect fax tone so no need for fax > detection on asterisk. > > Voice calls between the two work fine. > > But on a Fax call, I see this situation: > > A <=> Asterisk <=> B > > A: INVITE + Audio SDP => Asterisk => (same SDP) => B > > B: 200 OK + Audio SDP => Asterisk => (same SDP) => A > > * B Detects Fax-Tone! > > B: Re-Invite + UDPTL => Asterisk => (same SDP) => A > > A: 200 OK + UDPTL => Asterisk => 488 => B > > I tweakted the udptl setting in various ways, but I am unable to figure > out, why Asterisk is sending a 488 to B, after it first happily > forwarded the SDP to A and got confirmation from A it was happy to > accept that DSP. > > You could enable core debug and see if there's any insight, otherwise you'd have to actually provide the full traces. Asterisk also doesn't forward SDPs between sides so they're not the same SDP. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP address learning and timing problem
I don't know in that specific output what happened. Your best course of
action is to add further logging or step through the logic with all of the
knowledge you have of the RTP streams to understand what is happening.
On Mon, Apr 17, 2023 at 8:52 PM David Cunningham
wrote:
> Hi Joshua,
>
> Thank you for that. From the code it kind of looks like
> STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum:
>
> if (!ast_sockaddr_isnull(>strict_rtp_address)
> && STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(),
> rtp->rtp_source_learn.start)) {
> ast_verb(4, "%p -- Strict RTP learning complete - Locking on source
> address %s\n",
>
> Our call shows:
>
> # grep C-00024cd5 full.log | egrep 'Strict RTP'
> [Feb 22 11:16:41] VERBOSE[29023][C-00024cd5] res_rtp_asterisk.c:>
> 0x2b308c074f80 -- Strict RTP learning after remote address set to:
> xx.xx.154.111:18578
> [Feb 22 11:17:00] VERBOSE[29023][C-00024cd5] res_rtp_asterisk.c:>
> 0x2b315c01cbc0 -- Strict RTP learning after remote address set to:
> xx.xx.0.12:16498
> [Feb 22 11:17:00] VERBOSE[28191][C-00024cd5] res_rtp_asterisk.c:>
> 0x2b308c074f80 -- Strict RTP switching to RTP remote address
> xx.xx.154.111:18578 as source
> [Feb 22 11:17:00] VERBOSE[28191][C-00024cd5] res_rtp_asterisk.c:>
> 0x2b308c074f80 -- Strict RTP learning complete - Locking on source address
> xx.xx.154.111:18578
> [Feb 22 11:17:00] VERBOSE[28194][C-00024cd5] res_rtp_asterisk.c:>
> 0x2b315c01cbc0 -- Strict RTP switching source address to xx.xx.114.237:16498
> [Feb 22 11:17:01] VERBOSE[28194][C-00024cd5] res_rtp_asterisk.c:>
> 0x2b315c01cbc0 -- Strict RTP learning complete - Locking on source address
> xx.xx.114.237:16498
>
> I'm a bit confused because the second "Strict RTP learning after remote
> address set" should reset the rtp_source_learn.start timestamp, and yet the
> "Strict RTP learning complete" messages are less than 5000ms after that.
> What could be happening?
>
> Thanks again.
>
>
> On Tue, 18 Apr 2023 at 10:40, Joshua C. Colp wrote:
>
>> It's probably best if you read the logic[1]. There's an entire comment
>> that talks about how it works.
>>
>> [1]
>> https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158
>>
>> On Mon, Apr 17, 2023 at 7:10 PM David Cunningham <
>> dcunning...@voisonics.com> wrote:
>>
>>> Hi Joshua,
>>>
>>> Could you confirm if the 5 second period for learning a new audio stream
>>> is a minimum or a maximum? The unusual call flow in question results in
>>> Asterisk learning a new audio stream when we don't want it to, and having a
>>> minimum of say 2 seconds of audio would help avoid this.
>>>
>>> Thank you!
>>>
>>>
>>> On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp wrote:
>>>
>>>> On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp
>>>> wrote:
>>>>
>>>>> On Tue, Feb 28, 2023 at 9:50 AM David Cunningham <
>>>>> dcunning...@voisonics.com> wrote:
>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> Does anyone know if one of the "strictrtp" options disables RTP
>>>>>> learning? As far as I can tell from the documentation the values "no" and
>>>>>> "seqno" are more permissive in allowing other sources rather than less,
>>>>>> but
>>>>>> I thought I'd check.
>>>>>>
>>>>>
>>>>> Setting it to "no" disables the learning.
>>>>>
>>>>
>>>> Since I haven't gotten the email yet I'll just reply to my own.
>>>>
>>>> The "no" option disables strict RTP protection. Learning is part of
>>>> strict RTP protection, it is what determines what the source of media is
>>>> and then blocks other packets. There is no ability to set it
>>>> per-peer/per-endpoint.
>>>>
>>>> --
>>>> Joshua C. Colp
>>>> Asterisk Project Lead
>>>> Sangoma Technologies
>>>> Check us out at www.sangoma.com and www.asterisk.org
>>>> --
>>>> _
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>
>>>> Check out the new Asterisk community forum at:
>>>> https://community.asterisk.org/
>>>>
>>>> New to Asterisk? Start here:
>>>>
Re: [asterisk-users] RTP address learning and timing problem
It's probably best if you read the logic[1]. There's an entire comment that talks about how it works. [1] https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158 On Mon, Apr 17, 2023 at 7:10 PM David Cunningham wrote: > Hi Joshua, > > Could you confirm if the 5 second period for learning a new audio stream > is a minimum or a maximum? The unusual call flow in question results in > Asterisk learning a new audio stream when we don't want it to, and having a > minimum of say 2 seconds of audio would help avoid this. > > Thank you! > > > On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp wrote: > >> On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp wrote: >> >>> On Tue, Feb 28, 2023 at 9:50 AM David Cunningham < >>> dcunning...@voisonics.com> wrote: >>> >>>> Hello, >>>> >>>> Does anyone know if one of the "strictrtp" options disables RTP >>>> learning? As far as I can tell from the documentation the values "no" and >>>> "seqno" are more permissive in allowing other sources rather than less, but >>>> I thought I'd check. >>>> >>> >>> Setting it to "no" disables the learning. >>> >> >> Since I haven't gotten the email yet I'll just reply to my own. >> >> The "no" option disables strict RTP protection. Learning is part of >> strict RTP protection, it is what determines what the source of media is >> and then blocks other packets. There is no ability to set it >> per-peer/per-endpoint. >> >> -- >> Joshua C. Colp >> Asterisk Project Lead >> Sangoma Technologies >> Check us out at www.sangoma.com and www.asterisk.org >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > David Cunningham, Voisonics Limited > http://voisonics.com/ > USA: +1 213 221 1092 > New Zealand: +64 (0)28 2558 3782 > -- > _________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] couldn't allocate a port for RTP instance
On Fri, Apr 14, 2023 at 6:30 AM Fourhundred Thecat <400the...@gmx.ch> wrote:
>
> thank you, I found the problem:
>
> I had this in extensions.conf:
>
>same => n,GotoIf($[ "${DIALSTATUS}" = "BUSY" ]?6:7)
>
> and then I added one line above that, and forgot to adjust the numbers.
> So basically, the rule was going in an cycle.
>
> Is there a better way than to use hardcoded numbers for GotoIf ?
>
Yes. Use labels[1][2][3].
[1]
https://wiki.asterisk.org/wiki/display/AST/Goto+Application+and+Priority+Labels
[2]
https://wiki.asterisk.org/wiki/display/AST/Contexts,+Extensions,+and+Priorities#Contexts,Extensions,andPriorities-Applicationcalls
[3]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Application_GotoIf
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Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
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Re: [asterisk-users] couldn't allocate a port for RTP instance
On Fri, Apr 14, 2023 at 5:44 AM Fourhundred Thecat <400the...@gmx.ch> wrote: > Hello, > > my logs are flooded with: > > WARNING: The 'stasis/m:cdr:aggregator-0005' task processor queue > reached 5000 scheduled tasks again. > This means there's a ton of channel events being created. > > and then, when call came, I got this: > > ERROR: Oh dear... we couldn't allocate a port for RTP instance > '0x6e1e680fd670' > This means that there are a ton of channels or sessions up that are using all the RTP resources. > > WARNING: Unable to cancel schedule ID 0. This is probably a bug > (res_rtp_asterisk.c: dtls_srtp_stop_timeout_timer, line 2873). > > any idea what is happening, or how to troubleshoot his ? > There is insufficient information. You'd need to state what version of Asterisk, as well as which channel driver to begin with. Showing a console log may also provide information. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting PJSIP header from AMI
On Mon, Apr 10, 2023 at 8:25 PM Alex Zarubin wrote:
> Hello,
>
>
>
> We are moving from an older asterisk/SIP to a newer (18+) asterisk/PJSIP
> and trying to figure out how to add [identity] header when originating a
> call from AMI/PAMI.
>
> In the older version we would just set a variable like this:
>
>
>
> $action = new OriginateAction("SIP/….”);
>
> $action->setVariable('__SIPADDHEADER51',"Identity:
> $identity"); // $identity contains generated by 3rd party
> header
>
>
>
> Is there anything similar for
>
>
>
> $action = new OriginateAction("PJSIP/….”);
>
> ???
>
>
>
> that would work for PJSIP?
>
Yes, the PJSIP_HEADER dialplan function[1].
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_PJSIP_HEADER
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Joshua C. Colp
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Sangoma Technologies
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Re: [asterisk-users] 401 error
That's the extent of my vague memories of chan_sip then, someone else may be able to answer. On Fri, Mar 10, 2023 at 11:05 AM Jerry Geis wrote: > > > On Fri, Mar 10, 2023 at 9:49 AM Jerry Geis wrote: > >> >> >> On Thu, Mar 9, 2023 at 9:42 PM Jerry Geis wrote: >> >>> I have a SIP trunk - calls going out work fine. >>> >>> Trying to setup an incoming call with a DNIS >>> >>> When I dial the number - I see nothing on the CLI. >>> The person says the server is returning 401 >>> >>> How do I debug that. Using asterisk 18.8.0 >>> >>> Thanks >>> >>> Jerry >>> >> >> Thanks I am using chan_sip. Turning on "sip set debug on" I do se it. >> >> >> >> Using INVITE request as basis request - >> 0gQAAC8WAAACBAAALxYAAJJTfs5d3xOERBazRh4eM17uDp4m3CsM7ZnaEtaSRzwP@IP >> Found peer 'JJ' for 'phone' from IP:5060 >> >> <--- Reliably Transmitting (no NAT) to IP:5060 ---> >> SIP/2.0 401 Unauthorized^M >> Via: SIP/2.0/UDP >> IP:5060;branch=z9hG4bK+13778ac6c6ac51ac853b3fb4a387e5b71+sip+3+ae5ab1a2;received=IP^M >> From: "Caller" ;tag=IP+3+67d18b6f+9e6ad02d^M >> To: ;tag=as128621a0^M >> Call-ID: >> 0gQAAC8WAAACBAAALxYAAJJTfs5d3xOERBazRh4eM17uDp4m3CsM7ZnaEtaSRzwP@IP^M >> CSeq: 503124310 INVITE^M >> Server: Asterisk PBX 18.14.0^M >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE^M >> Supported: replaces, timer^M >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", >> nonce="6cbb5c2f"^M >> Content-Length: 0^M >> >> I dont see a reason why it failed. >> I tried nat=yes, made no difference. >> I tried insecure=very, made no difference. >> >> I do have: >> externip=X >> localnet=Y >> localnet=Z >> set in sip.conf >> >> As I mentioned - I can call out over this SIP trunk. >> What next ? >> Jerry >> > > > Just added insecure=very again, stopped and started. > > > [JJ] > type=friend > dtmfmode=rfc2833 > secret=yes > username=NUMBER > defaultuser=NUMBER > disallow=all > allow=ulaw > allow=alaw > context=smvoice-incoming > host=dnsname > canreinvite=yes > qualify=yes > insecure=very > > Got the same 401. > Thanks > > Jerry > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 401 error
On Fri, Mar 10, 2023 at 10:50 AM Jerry Geis wrote: > > > On Thu, Mar 9, 2023 at 9:42 PM Jerry Geis wrote: > >> I have a SIP trunk - calls going out work fine. >> >> Trying to setup an incoming call with a DNIS >> >> When I dial the number - I see nothing on the CLI. >> The person says the server is returning 401 >> >> How do I debug that. Using asterisk 18.8.0 >> >> Thanks >> >> Jerry >> > > Thanks I am using chan_sip. Turning on "sip set debug on" I do se it. > > > > Using INVITE request as basis request - > 0gQAAC8WAAACBAAALxYAAJJTfs5d3xOERBazRh4eM17uDp4m3CsM7ZnaEtaSRzwP@IP > Found peer 'JJ' for 'phone' from IP:5060 > > <--- Reliably Transmitting (no NAT) to IP:5060 ---> > SIP/2.0 401 Unauthorized^M > Via: SIP/2.0/UDP > IP:5060;branch=z9hG4bK+13778ac6c6ac51ac853b3fb4a387e5b71+sip+3+ae5ab1a2;received=IP^M > From: "Caller" ;tag=IP+3+67d18b6f+9e6ad02d^M > To: ;tag=as128621a0^M > Call-ID: > 0gQAAC8WAAACBAAALxYAAJJTfs5d3xOERBazRh4eM17uDp4m3CsM7ZnaEtaSRzwP@IP^M > CSeq: 503124310 INVITE^M > Server: Asterisk PBX 18.14.0^M > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE^M > Supported: replaces, timer^M > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", > nonce="6cbb5c2f"^M > Content-Length: 0^M > > I dont see a reason why it failed. > I tried nat=yes, made no difference. > I tried insecure=very, made no difference. > > I do have: > externip=X > localnet=Y > localnet=Z > set in sip.conf > > As I mentioned - I can call out over this SIP trunk. > What next ? > It matched peer 'JJ'. That peer would need to have insecure=very set, and chan_sip then reloaded. Providing the actual peer would also be faster for anyone to provide help. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 401 error
On Thu, Mar 9, 2023 at 10:43 PM Jerry Geis wrote: > I have a SIP trunk - calls going out work fine. > > Trying to setup an incoming call with a DNIS > > When I dial the number - I see nothing on the CLI. > The person says the server is returning 401 > > How do I debug that. Using asterisk 18.8.0 > There are two different SIP channel drivers. If using chan_sip then "sip set debug on" will show you the SIP traffic, if using chan_pjsip then "pjsip set logger on" will. After confirming it you then look at the configuration. You would need to ensure that you are matching the incoming traffic against either a peer for chan_sip (host= in a peer), or an endpoint in chan_pjsip (identify section). You'd also need to confirm that you haven't configured it to challenge those calls for authentication (insecure=very in chan_sip, and not having auth or inbound_auth set on endpoint in chan_pjsip). -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 5s delays before executing the dialplan
On Wed, Mar 1, 2023 at 6:49 PM Kingsley Tart wrote: > On Tue, 2023-02-28 at 09:50 -0400, Joshua C. Colp wrote: > > Is the local hostname configured in /etc/hosts and not reliant on an > outside DNS server? Are you using ICE or STUN at all? > > > Hi, > > thanks for responding. > > No ICE or STUN. > > Some of the servers have entries for themselves in /etc/hosts and some > rely on external DNS. That is not by design, it's just how it happened and > I shall sort that out. > > I can't figure out how whether the delays were on only the ones without > /etc/hosts entries, but I can be sure that some without those entries > definitely did experience those delays. > > Does 5 seconds match some sort of DNS timeout within Asterisk that could > have been the cause? > Within Asterisk, no. It uses system level stuff to do the DNS resolution, that has its own timeouts that Asterisk doesn't control. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP address learning and timing problem
On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp wrote: > On Tue, Feb 28, 2023 at 9:50 AM David Cunningham < > dcunning...@voisonics.com> wrote: > >> Hello, >> >> Does anyone know if one of the "strictrtp" options disables RTP learning? >> As far as I can tell from the documentation the values "no" and "seqno" are >> more permissive in allowing other sources rather than less, but I thought >> I'd check. >> > > Setting it to "no" disables the learning. > Since I haven't gotten the email yet I'll just reply to my own. The "no" option disables strict RTP protection. Learning is part of strict RTP protection, it is what determines what the source of media is and then blocks other packets. There is no ability to set it per-peer/per-endpoint. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mailing Lists
Greetings, As some of you noticed (and likely some didn't) the mailing lists were down for a period of time at the start of this year, and over the past week. We believe we've stabilized things to allow them to continue to run and will continue to monitor. Sometime this year I will explore moving the mailing lists to a different solution for those who still prefer mailing lists, which should also improve deliverability. When this begins and occurs I will start discussions on the existing mailing lists. For user facing questions I do urge people to use the community forums[1] which have effectively taken over for discussions. They are much more active with both questions and answers, and are easier to search and find information on. You can set it up to behave as a mailing list if you want including replying using email. Cheers, [1] https://community.asterisk.org/ -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP address learning and timing problem
On Tue, Feb 28, 2023 at 9:50 AM David Cunningham wrote: > Hello, > > Does anyone know if one of the "strictrtp" options disables RTP learning? > As far as I can tell from the documentation the values "no" and "seqno" are > more permissive in allowing other sources rather than less, but I thought > I'd check. > Setting it to "no" disables the learning. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 5s delays before executing the dialplan
On Tue, Feb 28, 2023 at 9:48 AM Kingsley Tart wrote: > Hi, > > We've recently hit an issue with Asterisk 18.8.0 where a call comes in > via SIP (using pjsip) but it can take 5 seconds before starting to > execute the dialplan. > > This was intermittent, but frequent (eg approx half of the calls). > > We have verbose logging on, but I didn't see any errors. > > Running asterisk -r - and then watching SIP traffic in another > window showed the INVITE coming in, then a good 5 seconds before > dialplan execution started showing within the Asterisk console. > > We've never seen this before, but it affected 6 our of 8 of our > Asterisk servers. They're running in debian 11 VMS, with the VMs > running under KVM on the host OS which is also debian 11. > > Any suggestions where to look next? We've been running Asterisk for > years but never seen this issue before. > > FWIW, the match= lines for the SIP proxies sending to Asterisk are > configured by IPv4 address, not host name. > Is the local hostname configured in /etc/hosts and not reliant on an outside DNS server? Are you using ICE or STUN at all? -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [External] Asterisk rtp.conf stunaddr setting - what happens if there is an outage
On Tue, Feb 7, 2023 at 11:18 AM Dan Cropp wrote: > Thank you Joshua. > > > > Going back to your idea of the ice_host_candidates. (Again, apologize for > my ignorance on networking). > > Do I understand correctly? We could use this formula for systems that have > no one accessing the (where 192.168.1.10 is the internal IP) and 1.2.3.4 is > the NAT’s public IP for Asterisk? > > > > 192.168.1.10 => 1.2.3.4,include_local_address > > > > Using this, would we no longer need the stunaddr configured? > You don't need the include_local_address option but otherwise yes. This will cause the ICE host candidates to be 1.2.3.4 instead of the local IP address 192.168.1.10 removing the need to use STUN to discover the public IP address. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk rtp.conf stunaddr setting - what happens if there is an outage
On Mon, Feb 6, 2023 at 6:05 PM Dan Cropp wrote: > A quick follow-up. > > > > Looking at other customers running 18.12.1 who reported problems at the > exact same time with AWS issue described below. > > > > We are seeing similar behavior. > > For these systems, the third STUN failure occurs. We were able to answer > the call because the SIP provider didn’t CANCEL the call. > > However, upstream from the service provider the calls were terminated. > > Resulting in a call from the SIP provider to Asterisk that’s live, but > there is no caller so it appears to be dead air. > > > > Does the res_rtp_asterisk stunaddr DNS TTL expiration mentioned in change > ID I7955a046293f913ba121bbd82153b04439e3465f require the dnsmgr.conf to be > enabled? > It doesn't use dnsmgr so it's not required to be enabled. If the TTL is long, or it's cached locally then it could stick around longer. Fundamentally though is there a reason you're using STUN in the first place? Can you not just configure the public IP address and not rely on an external STUN server? rtp.conf has ice_host_candidates specifically for situations like AWS. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] set codec based on B side
On Tue, Jan 31, 2023 at 4:14 PM Fabian Borot wrote: > > Using Asterisk 18.12.0, a little confused on how to configure my > pjsip.conf file to determine the codec to use for a call > > I have 2 endpoints: > [Alice] > disallow:all > allow:ulaw,alaw,g729 > > [Bob] > disallow:all > allow:ulaw,alaw,g729 > > Alice calls into Asterisk on ext 100 and then we dial Bob > I want to wait until Bod side codec is chosen to answer Alice and have > each channel use the codec chose on Bob side. > That is not currently possible. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a list of Channel ARI requests that are allowed when the call is not handed off to the Stasis application
On Mon, Jan 30, 2023 at 7:30 PM Dan Cropp wrote:
> We have used AMI for many years and I’m in the process of migrating to ARI.
>
>
>
> My understanding is the call should be handed off to Stasis for the ARI
> application to control it.
>
>
>
> I was playing around with things and discovered the ARI hangup (DELETE
> /channels/{channelId}) allowed me to hangup calls even when no StasisStart
> is received.
>
> I tried some other requests and they did not seem to work. This is what I
> expected to happen for the hangup.
>
> Are there other commands that are allowed on channels when the call is not
> in the Stasis app? (Obviously creating a channel and externalMedia will
> work because they create new channels).
>
There's not really a list, some just work due to the internal way they work
in Asterisk.
>
>
> Also, to be fault tolerant, I noticed a call handed off to Stasis app will
> remain in the Stasis app, even if the ARI/WebSocket connection drops (power
> outage, etc). When establishing the ARI/WebSocket connection, the first
> thing I am planning to do is GET a list of the channels. This returns all
> of the channels in the system and not just the channels that are in this
> Stasis apps control. I plan to go through the list and identify the
> channels dialplan data. Look for app_name of Stasis and the app_data
> (comma-delimited).
>
> If app_name = “Stasis” and app_data’s first section of the comma-delimited
> parse portion matches the Stasis app name this instance is used, I take
> control of this channel.
>
> I am planning this additional check because I noticed the Stasis power
> outage scenario resulted in channels stuck in the Stasis app. If I don’t
> take control of these channels, it’s possible to eventually have
> hundreds/thousands of channels. For SIP calls, the other end eventually
> hangs up. However, this isn’t the case with Local channels. Particularly
> when both ends are locally controlled by Stasis.
>
>
>
> Does this sound like I am on the right track for migrating from AMI to
> Stasis, ARI/Websocket support?
>
You may be able to get the application details[1][2] which would tell you
what the application is subscribed to, which would include the channels.
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Applications+REST+API#Asterisk20ApplicationsRESTAPI-get
[2]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+REST+Data+Models#Asterisk20RESTDataModels-Application
--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Testing
This is just a test of the asterisk-users mailing list. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PlayBack
On Wed, Dec 14, 2022 at 8:02 AM wrote: > Hi, > > not installed > pbx.c:2907 pbx_extension_helper: No application 'ControlPlayback' for > extension > > My version or Linux versionto old? I installed Asterisk using #aptitude > installed asterisk > Asterisk 16.2.1~dfsg-1+deb10u2 built by nobody @ buildd.debian.org on a > unknown running Linux on 2020-08-26 22:53:40 UTC > The module exists in 16.2.1, it is from the app_controlplayback module. The project doesn't build those packages, so I don't know how they were built. I also don't know your configuration (you might have modules.conf set to only load specific things, in which case you'd need to also load app_controlplayback). -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PlayBack
On Wed, Dec 14, 2022 at 7:32 AM wrote: > Hi, > Iam using Playback do play an sound file. > Is there a programm where I can move forward an backward within the > sound file with DTMF tones. > The ControlPlayback dialplan application[1] allows this. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Application_ControlPlayback -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.12.1 to 18.15.0 upgrade seems to have introduced a behavior where PJSIP is unable to send a response to OPTIONS (seems to resolve after anywhere a period of time)
On Wed, Dec 7, 2022 at 11:34 AM Joshua C. Colp wrote: > On Wed, Dec 7, 2022 at 11:26 AM Dan Cropp wrote: > >> On two VMs, we encounter a strange behavior when we upgrade from 18.12.1 >> to 18.15.0 (also tried 18.15.1 last night). >> >> When we roll the VMs back to 18.12.1, we don’t see the behavior repeat. >> >> >> >> We have a Kamailio VM front ending the asterisk. >> >> >> >> It sends OPTIONS messages periodically. >> >> >> >> After startup (and also after reloading configuration settings), we see >> periods where the response can’t be sent. >> >> After a period of time, it suddenly starts working. >> >> >> > I haven't seen this before, and haven't seen any other reports of it so > far. The OPTIONS code itself hasn't changed between the two. There was a > fix for a crash in send_stateful_response so adding log messages to the > error cases is likely needed to see in particular which one is failing. > > Ha, those changes haven't even landed yet. It's pretty much a thin wrapper over PJSIP stuff in 18.15.1. The PJSIP versions are also fairly close too. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.12.1 to 18.15.0 upgrade seems to have introduced a behavior where PJSIP is unable to send a response to OPTIONS (seems to resolve after anywhere a period of time)
0f.4d4a0332.0 > > Call-ID: 5ff0944d46b1692c-2858982@192.168.12.10 > > From: > sip:kamailio@192.168.12.10;tag=f1bcd6806a18022e516c4139c95990f1-92130971 > > To: > sip:box_b@192.168.12.120;tag=z9hG4bK10f.4d4a0332.0 > > CSeq: 10 OPTIONS > > Accept: application/sdp, application/dialog-info+xml, > application/simple-message-summary, application/xpidf+xml, > application/cpim-pidf+xml, application/pidf+xml, application/pidf+xml, > application/dialog-info+xml, application/simple-message-summary, > message/sipfrag;version=2.0 > > Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, > CANCEL, UPDATE, PRACK, MESSAGE, REFER > > Supported: 100rel, timer, replaces, norefersub > > Accept-Encoding: identity > > Accept-Language: en > > Server: Asterisk PBX 18.15.1 > > Content-Length: 0 > > > > Portion of the pjsip.conf settings… > > > > [Kamailio] > > type = aor > > authenticate_qualify = yes > > contact = sip:192.168.10.235 > > ;outbound_proxy=sip:192.168.12.10 > > > > [identify158] > > type = identify > > endpoint = Kamailio > > match = 192.168.12.10 > > > > [Kamailio] > > type = endpoint > > context = IS > > transport = transport1 > > aors = Kamailio > > accountcode = 29 > > dtmf_mode = inband > > device_state_busy_at = 48 > > force_rport = no > > moh_passthrough = no > > identify_by = username,ip,header > > disallow = all > > allow = ulaw > > asymmetric_rtp_codec = yes > > acl = acl6 > > outbound_proxy=sip:192.168.12.10 > > > > Was there some configuration change introduced after 18.12.1 that I missed? > Any such things would be in the upgrade notes, but no. > > > Any thoughts? > I haven't seen this before, and haven't seen any other reports of it so far. The OPTIONS code itself hasn't changed between the two. There was a fix for a crash in send_stateful_response so adding log messages to the error cases is likely needed to see in particular which one is failing. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cannot load res_geolocation.so
On Mon, Dec 5, 2022 at 4:31 PM Nick Olsen wrote: > Hello, > > On a fresh install of 18.9 Cert2 (Or the latest 19 if I recall the > previous version I tried. > > PJSIP fails to load properly. It seems that the new res_geolocation module > fails to load. But I can't seem to figure out why. And being that it's a > fairly new module (So it seems) google-fo isn't being very helpful. This is > running on Debian 11 and a freshly compiled install with only "make > samples" done to create the config files. Any help would be appreciated! > > newasterisk*CLI> module load res_pjsip.so > Unable to load module res_pjsip.so > Command 'module load res_pjsip.so ' failed. > [Dec 5 15:26:18] ERROR[2420]: loader.c:283 module_load_error: res_pjsip > loaded before dependency res_geolocation! > > newasterisk*CLI> module load res_geolocation.so > Unable to load module res_geolocation.so > Command 'module load res_geolocation.so ' failed. > [Dec 5 15:26:28] WARNING[2420]: config_options.c:1102 > xmldoc_update_config_type: Cannot update type 'location' in module > 'res_geolocation' because it has no existing documentation! > [Dec 5 15:26:28] ERROR[2420]: res_geolocation/geoloc_config.c:672 > geoloc_config_load: Failed to register geoloc location object with sorcery > This would mean that the documentation isn't in the core-en_US.xml file, normally located in the /var/lib/asterisk/documentation directory. I just built 18.9-cert3 and it is definitely there for me: jcolp@kappa:~/development/asterisk/public [certified/18.9-cert3| …2⚑ 4]> grep "geolocation" /var/lib/asterisk/documentation/core-en_US.xml res_geolocation res_geolocation Get or Set a field in a geolocation profile This geolocation profile will be applied to all calls received This geolocation profile will be applied to all calls received And the module loads fine: *CLI> module show like geolocation Module Description Use Count Status Support Level res_geolocation.so res_geolocation Module for Asterisk 2 Running core res_pjsip_geolocation.so res_pjsip_geolocation Module for Asteris 0 Running core 2 modules loaded Did you build Asterisk putting things in other directory locations? Is there an old core-en_US.xml file somewhere? -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk unable to do DNS lookups
On Wed, Nov 30, 2022 at 6:03 PM TTT wrote: > I’ve noticed on several occasions that if Asterisk starts without a > network connection, then even if the network connection is restored, DNS > lookups fail. > > > > After the connection is restored I can successfully do NSLOOKUPs from the > command line, but the IAX2 registration attempts keep failing because > Asterisk has a problem. > > > > My questions are: > > 1. Is there a way to make Asterisk update (whatever is wrong) and > resume successful DNS lookups? > There is dnsmgr which is configured in dnsmgr.conf that will periodically refresh. Older modules (chan_iax2 and chan_sip) don't do lookups at use time, but instead do it once and store the result (unless they support dnsmgr and dnsmgr updates it). > 2. Is there a way from the Asterisk CLI to detect when Asterisk > enters this state of failed DNS lookups? (Other than tracking IAX2/SIP > registration failures) > Not really. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] possibility to cancel call duration limit set in app Dial with options S(x) or L(x:y:z) during a call
On Wed, Nov 30, 2022 at 5:57 AM Nenad Radosavljevic wrote:
> Hello everyone,
>
> Does anyone know is it possible to cancel the call duration limit set in
> app Dial with options S(x) or L(x[:y[:z]]), by for instance entering custom
> feature code (application map) during the call ?
>
> I have read somewhere that bridge features can be set using the
> ${BRIDGE_FEATURES} variable on channel, but it was not clear to me can they
> be modified that way after the bridge has started ?
>
> Idea is to make application map for caller channel that will do
> Set(BRIDGE_FEATURES=S(0)) on its invocation via feature code (with some
> "call unlimit autorization" dialplan logic before it), in order to be able
> to un-limit call duration on-demand.
>
The Dial options are not dynamic during the call, they're just read and set
when Dial() is invoked. The same applies for BRIDGE_FEATURES. The TIMEOUT
dialplan function[1] is just what can be changed during the call.
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_TIMEOUT
--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
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Re: [asterisk-users] MixMonitor not recording through transfer
On Tue, Nov 29, 2022 at 9:24 PM Carlos Chavez wrote: > I have the following scenario: > > Agent calls external number > > Mixmonitor starts recording call > > After agent speaks with customer they need to transfer them to an > extension that will simply play a message > > Customer hangs up > > The problem is that the recording stops the moment the agent > transfers the call to the other extension. We need the recording to > include the message from the other extension. We use Asterisk 16 on > this server. I know that the AUDIOHOOK_INHERIT function was deprecated > long ago so I should not need anything extra to keep recording through > transfers, or am I wrong? > MixMonitor records the channel it is placed on, and follows the channel it is placed on. If you're recording the AGENT channel then it won't follow. You have to record the customer channel. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cps limit of asterisk
The only answer is the one you benchmark yourself for your environment, deployment, and usage. On Wed, Nov 23, 2022 at 3:02 PM Tahir Almas Dhesi wrote: > What is maximum cps limit of a good asterisk server (single node ) ? > > regards > *Tahir Almas* > > Managing Partner > ICT Innovations > http://www.ictinnovations.com > Leveraging open source in ICT > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Force voicemail check / MWI generation
On Fri, Nov 18, 2022 at 9:53 AM Nick Olsen wrote: > Hello, > > I've got a handful of servers running asterisk 16 currently with voicemail > stored in a database via ODBC. > > Some users register to multiple servers and I'm having an issue with MWI. > > Specifically Polycom phones seem to not be able to use two different MWI > sources. So in my immediate instance, a voicemail is left on the primary > server. The primary server generates the requisite notify and properly sets > the phone's MWI light and makes an audible MWI sound. > > The phone then re-registers to the secondary instance at some point. Even > though these machines share a common voicemail database, asterisk isn't > aware that a voicemail now exists in the database and thus tells the phone > there are no messages. Clearing the phones MWI indications. The phone then > registers to the primary again rinse/repeat. > > Is there a way to force asterisk to somehow get an update when a voicemail > is dropped in the database? If I manually delete a voicemail from the > database via SQL commands asterisk does seem to pick it up shortly > thereafter. But it doesn't seem to learn that new voicemails are available. > Any help would be greatly appreciated. Thank you! > There is polling functionality[1] which may work. It's not instant, it will update based on the frequency. [1] https://github.com/asterisk/asterisk/blob/18/configs/samples/voicemail.conf.sample#L190 -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answer()ing a local Originate takes 500ms!?
On Fri, Nov 11, 2022 at 12:15 PM Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Friday 11 November 2022 at 17:11:26, Joshua C. Colp wrote: > > > On Fri, Nov 11, 2022 at 12:09 PM Antony Stone wrote: > > > > > > https://wiki.asterisk.org/wiki/display/AST/Application_Answer tells me > > > that the Answer() application takes an optional parameter which causes > > > Asterisk to wait that number of milliseconds before returning to the > > > dialplan after answering the call. > > > > > > Does this undocumentedly default to 500? > > > There is a hard coded minimum of 500 milliseconds for media to flow. > You'd > > have to modify the code to remove it. > > Urgh! > > Is there _anything_ I can do in either the Originate() or the Answer() to > avoid this, without having to rebuild Asterisk? > Not really, it waits until media flows. That's the way it is written. > > And, separately, please can I request that: > > a) this minimum is documented > > b) it can be over-ridden at the user's own risk if the supplied parameter > is > lower than 500. > Any such things should go into a JIRA issue[1]. [1] https://issues.asterisk.org/jira -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answer()ing a local Originate takes 500ms!?
On Fri, Nov 11, 2022 at 12:09 PM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:
> Hi.
>
> Asterisk 16.2.1
>
> I have a dialplan where one context (named "inbound") performs:
>
> Originate(Local/${Target}@inOrig,exten,inbound,${EXTEN},208)
>
> The idea is that this command will spawn a "call" to the context "inOrig"
> on
> the same machine, and then return to the "inbound" context at priority 208.
>
> Priority 208 is simply a NoOp(Returned from inOrig)
>
> The "inOrig" context does:
>
> NoOp(Answering inbound call)
> Answer()
> NoOp(Returned to inbound context)
> Originate(Local/${EXTEN}@dialout,exten,BridgIt,${EXTEN},1)
>
> It's all doing what I want / expect, but I am seeing, completely
> consistently,
> a 500ms delay in the Answer() application.
>
> So, I get the following sequence of timings:
>
> 08:41:49.514918 inbound:201 Originate(.)
> 08:41:49.516459 inOrig:1 NoOp(Answering inbound call)
> 08:41:49.517016 inOrig:2 Answer()
> 08:41:49.517489 inbound:208 NoOp(Returned from inOrig)
> 08:41:50.017454 inOrig:3 NoOp(Returned to inbound context)
>
> I have analysed dozens of calls and there is always a ~500ms delay between
> when the Answer() has clearly completed (because control returns to
> priority
> 208 of the "inbound" context), and when the inOrig context continues with
> the
> following NoOp.
>
> https://wiki.asterisk.org/wiki/display/AST/Application_Answer tells me
> that
> the Answer() application takes an optional parameter which causes Asterisk
> to
> wait that number of milliseconds before returning to the dialplan after
> answering the call.
>
> Does this undocumentedly default to 500?
>
> Are the results I'm seeing expected, is there something wrong with my
> dialplans, is there some way to eliminate this delay?
>
There is a hard coded minimum of 500 milliseconds for media to flow. You'd
have to modify the code to remove it.
--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] CSeq reset on re-INVITE
On Sun, Oct 30, 2022 at 5:00 PM David Cunningham wrote: > Hi Joshua, > > Thanks very much. I presume this is the relevant part: > "strictly monotonically increasing and contiguous CSeq sequence numbers > (increasing-by-one) in each direction" > > In that case I wonder what could be causing the 404 Not Found error. I've > attached the relevant SIP packets from the Asterisk log. Can anyone see an > issue that would cause the error? > Based on the provided trace the signaling appears correct. I don't think there's anything on the Asterisk side wrong, and don't think it'll lead to an answer. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CSeq reset on re-INVITE
On Fri, Oct 28, 2022 at 6:28 PM David Cunningham wrote: > Hello, > > We have a problem where Asterisk is resetting the CSeq on a re-INVITE, and > the phone receiving the re-INVITE is rejecting it, probably as a result of > that. Would anyone be able to offer any insight please? > > The scenario is: > > Phone A makes call 1 to Asterisk which dials call 2 to phone B, which > answers the call. > > Phone B puts call 1 on hold, makes call 3 to Asterisk which dials call 4 > to phone C, which answers the call. > > Phone B does an attended REFER transfer of call 2 to call 3, taking itself > out of the call. Asterisk bridges the remaining calls, so phones A and C > are now talking to each other. > > Asterisk sends a re-INVITE to phone A with a P-Asserted-Identity, to tell > phone A the updated details of phone C that it's talking to. However phone > A rejects the re-INVITE with a "404 Not found" error. > > The only explanation I can see for the "404 Not found" error is that call > 1 was set up with "CSeq: 954698786 INVITE", whereas the re-INVITE Asterisk > sends with the P-Asserted-Identity has "CSeq: 102 INVITE". Why is Asterisk > resetting the CSeq on the re-INVITE, and doesn't this appear to be > incorrect? > It's not incorrect. Each direction has its own CSeq[1]. From Phone A to Asterisk can be 954698786 and from Asterisk to Phone A can be 102. [1] https://www.rfc-editor.org/rfc/rfc3261#section-12.2.1.1 -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP audio
On Tue, Oct 18, 2022 at 4:56 PM Jerry Geis wrote: > > > On Tue, Oct 18, 2022 at 3:38 PM Jerry Geis wrote: > >> Has there been issues where "once in a while" RTP audio does not work ? >> >> Example: connection to Cisco call manager - works mostly all the time. >> >> once in a great while - person does not hear the "beep" when calling in. >> once in a great while - person they hear the beep - but do not hear the >> audio public address. >> >> What would I be looking for to track this beast down ? >> >> This is my SIP trunk >> [LSVOIP] >> type=friend >> dtmfmode=rfc2833 >> secret=password >> username=LSVOIP >> defaultuser=LSVOIP >> disallow=all >> allow=ulaw >> allow=alaw >> context=incoming >> host=172.1.1.1 >> canreinvite=yes >> qualify=yes >> insecure=invite >> >> Thoughts? >> >> Jerry >> > > > Is there any kind of pjsip vs old SIP (which I am using) issue happening > here. (asterisk 18.14.0) > No. The media stack between the two is the same, and is the existing one that has existed for years. The starting point for any issue like this is a packet capture that you can examine in wireshark to see what media is flowing, if any, where, and the signaling. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] @-sign gets transmitted as %40 in outgoing SIP packets (CallerID)
On Mon, Oct 17, 2022 at 10:02 AM Markus wrote: > Hi list, > > I'm using Asterisk 11.25.0 and would like to set > "anonymous@anonymous.invalid" as outgoing caller ID via SIP: > > Set(CALLERID(num)=anonymous@anonymous.invalid) > > However, when I look at the outgoing packet with tcpdump I see that the > @ is not being transmitted correctly: > > From: ;tag=as26264e65 > > My question is how can I get Asterisk to send a "@" instead of "%40"? > The number is sent as the user portion in SIP. That can't have the "@" in it, as that is invalid and against spec - it gets turned into "%40". If what you ACTUALLY want is the user portion to be "anonymous" and the domain portion to be "anonymous.invalid" then you may be able to set the callerid presentation[1] to not be allowed: Set(CALLERID(pres)=prohib) Though I haven't tested it on chan_sip in 11. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CALLERID -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Muliticast not connecting
On Thu, Oct 13, 2022 at 2:32 PM Eric Wieling wrote: > > > On 10/13/22 13:25, Joshua C. Colp wrote: > > On Thu, Oct 13, 2022 at 2:16 PM Jerry Geis > <mailto:jerry.g...@gmail.com>> wrote: > > > > I have a simple dialplan with asterisk 18.14.0 > > > > exten => 141,1,Answer > > exten => 141,n,Noop(MC) > > exten => 141,n,Playback(beep) > > exten => 141,n,Dial(MulticastRTP/basic/239.168.4.90:30410//t(15) > > <http://239.168.4.90:30410//t(15)>) > > exten => 141,n,Hangup > > > > Most times this works just fine ... Once in a while the person hears > > the beep - but nothing connects on the multicast. > > > > What might this be? How can I tell what is happening and why it does > > not connect? > > > > is it valid to put : > > exten => 141,n,Dial(MulticastRTP/basic/239.168.4.90:30410//t(15) > > <http://239.168.4.90:30410//t(15)>) > > exten => 141,n,Dial(MulticastRTP/basic/239.168.4.90:30410//t(15) > > <http://239.168.4.90:30410//t(15)>) > > > > So if the first one doesnt connect perhaps the second one will ??? > > Thanks > > > > > > Multicast doesn't connect. There is no session. RTP is thrown out onto > > the network using multicast, and then devices pick it up. Asterisk has > > no idea what (if anything) is receiving it. You'd want to do a packet > > capture to see what is being multicast. > > > > Does this mean things like DIALSTATUS won't work as expected? > It'll always be considered answered. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Muliticast not connecting
On Thu, Oct 13, 2022 at 2:16 PM Jerry Geis wrote: > I have a simple dialplan with asterisk 18.14.0 > > exten => 141,1,Answer > exten => 141,n,Noop(MC) > exten => 141,n,Playback(beep) > exten => 141,n,Dial(MulticastRTP/basic/239.168.4.90:30410//t(15)) > exten => 141,n,Hangup > > Most times this works just fine ... Once in a while the person hears the > beep - but nothing connects on the multicast. > > What might this be? How can I tell what is happening and why it does not > connect? > > is it valid to put : > exten => 141,n,Dial(MulticastRTP/basic/239.168.4.90:30410//t(15)) > exten => 141,n,Dial(MulticastRTP/basic/239.168.4.90:30410//t(15)) > > So if the first one doesnt connect perhaps the second one will ??? > Thanks > Multicast doesn't connect. There is no session. RTP is thrown out onto the network using multicast, and then devices pick it up. Asterisk has no idea what (if anything) is receiving it. You'd want to do a packet capture to see what is being multicast. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying asterisk on AWS
On Thu, Oct 6, 2022 at 10:24 AM Jerry Geis wrote: > >The sample configuration file outlines how things work, and the options for > >it: > >https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L874 > >in general localnet and externip (or externaddr, or externhost) > > I added: > > externip=xxx > nat=force_rport,comedia > > to the general section of sip.conf > > its still sending to the local IP. > Look at the actual SIP/SDP signaling to see what is being sent and what IP addresses are being used. If they're correct, then see if you are receiving traffic using "rtp set debug on". If you aren't then it's something outside of Asterisk preventing incoming traffic. Until Asterisk receives traffic it can't know the IP address+port to send outgoing to, beyond what was given in the SIP/SDP. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
