similar to: SIP trunk between asterisk boxes

Hey all! I recently tried the dtmf_mode "auto_info" on my setup to support endpoints that only understand SIP INFO as a fallback. My setup is the following: Endpoint A (RFC4733) --> Asterisk <-- Endpoint B (SIP INFO) Both are configured with "auto_info" dtmf_mode in pjsip.conf. What I ran into is, that DTMF sent from endpoint A to endpoint B is additionally sent via

On Tue, Jul 11, 2023 at 3:40 PM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, Jul 11, 2023 at 3:38 PM TTT <lists at telium.io> 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

Hi list! ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF gets transmitted throughout the conference. I've tried Asterisk 10.7.1 from the official RPMs and 10.8.0 compiled from source. I've confirmed that it's disabled via the CLI "confbridge show profile user <profilename>". It's an all-SIP scenario with RFC2833 as the DTMF protocol.

Hi, it seems that there is a bug in asterisk's dtmf mode autodetection. Assume following sip.conf: [sipprovider] disallow=all allow=g726 dtmfmode=auto DTMF does not work. It seems rfc2833 mode is chosen despite it being obvious that this cannot work! The following configuration is necessary to get DTMF to work: dtmfmode=info In my opinion, this behaviour is counter-intuitive. I am using

On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, Feb 28, 2023 at 9:50 AM David Cunningham < > dcunningham at 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

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 <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Thank you for that. From the code it kind of looks like

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 <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Could you confirm if the 5 second period for learning a new audio stream > is a minimum

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(&rtp->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: #

Response below... On Fri, Dec 27, 2019 at 12:02 PM David P <davidswalkabout at gmail.com> wrote: > > > > > I'm looking for a way of detecting in my dialplan when a peer becomes > > non-responsive after answering. [deleted] Is there a way to configure > > a handler for this state? > > > > We use v14.7.6 and we dial the peer this way: > > >

Thanks for your (fast) reply ! Le ven. 18 janv. 2019 à 16:32, Joshua C. Colp <jcolp at digium.com> a écrit : > On Fri, Jan 18, 2019, at 11:22 AM, Olivier wrote: > > Hello, > > > > I've just read in [1] about SIP MESSAGE addition to both chan_pjsip and > > ConfBridge. > > It seems very interesting addition as it brings the capability to mix > >

greetings asterisk users :) ive just deployed version 17 and migrated as best I can to pjsip. I can receive calls, and get to my mailbox prompt, however placing calls seems impossible with the following error on dial: Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid = 517890) dunkel*CLI> dunkel*CLI> == Setting global variable 'SIPDOMAIN' to

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 <jcolp at sangoma.com> wrote: > On

Hello, Where can I find doc about PJSIP's ice_support parameter ? Do you need to configure things elsewhere in Asterisk config files (rtp.conf, PJSIP transport sections, ...) to make ICE work properly ? I'm asking because, if I'm not mistaken, STUN requires setting a STUN server so I think ICE most probably, should also require settings some public resources. Best regards

On 13.07.20 at 00:17 Joshua C. Colp wrote: > On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins <dan at nimblea.pe> wrote: > >> Asterisk 18 will have support based on this asterisk update Matt F did for >> CommCon's sponsor slots >> >> https://youtu.be/eas1csaX-wc >> >> > As well support will go into Asterisk 16 and 17 as well. It's just been

I deleted the res_resolver_unbound.so module, and now it works as expected. So, the problem is related to the 'unbound' resolver? FYI: I'm using Asterisk 16.2 installed from Debian 10 repository. Best regards, Leonid Fainshtein On Thu, Aug 27, 2020 at 3:01 PM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Thu, Aug 27, 2020 at 8:58 AM Leonid Fainshtein < >

Hello list, Hope you are all doing fine! I have stumbled over some piece of dialplan code in which apparently they were trying to avoid recording the DTMF tones in the wav file. It is really messy and I am not sure if this really works. So after a bit of research I found this comment ( https://community.asterisk.org/t/asterisk-dtmf-record/65040) in which it is said: *"Asterisk strips the

examples of "interesting" information like ICE result and howto make "minimal" configuration of pjproject.conf i.e. for  debugging app_queue.so core set debug 5 app_queue.so for debugging RTP core set debug 10 rtp_engine core set debug 10 res_rtp_asterisk rtp set debug on logger.conf rtp => debug,verbose(5) so i mean in

On Mon, Jun 29, 2020 at 6:46 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Sun, Jun 28, 2020 at 2:26 PM Dovid Bender <dovid at telecurve.com> wrote: > >> Hi, >> >> We have a box up and we are starting to see a lot of "Exceptionally long >> queue length queuing" in the logs. From all the research so far it seems >> like this leads to

Hello. I am having a lot of problems with SIP through NAT. So, I decided to adopt PJSIP. However, I am not able to make the extensions ring when receiving a call from the queue. I'm using telnet to include the extension and on the asterisk console, it even shows Called PJSIP/6001, but the extension doesn't ring. If I call from extension to extension, it works normally. telenet:

Thanks for your responses - it looks like I have the following options, in order of ease: 1: Modify and recompile app_record.c Change line 471 https://github.com/asterisk/asterisk/blob/master/apps/app_record.c#L471 from status_response = "DTMF"; to status_response = dtmf_integer; Pro: Free, easy Con: Have to remember to edit module each time a new Asterisk update comes out 2: