Autoanswer second call not being answered

[cody-herbst]

I'm trying to set up baresip so that it immediately answers calls. Ive configured multiple user-agents with all having roughly the same configuration. I'm seeing the first call being correctly autoanswered but the second (even though it is configured to autoanswer) prompts me to answer the call. Am I missing something? Here are my configurations.

baresip v4.7.0

accounts:

<sip:officers@0.0.0.0:5060>;regint=1;answermode=auto;transport=udp;audio_player=alsa,plughw:0,0,0;audio_source=alsa,plughw:0,0,0;ausrc_srate=8000
<sip:crew@0.0.0.0:5060>;regint=1;answermode=auto;transport=udp
<sip:topside@0.0.0.0:5060>;regint=1;answermode=auto;transport=udp;audio_player=alsa,plughw:0,0,2;audio_source=alsa,plughw:0,0,2;ausrc_srate=8000
<sip:engineering@0.0.0.0:5060>;regint=1;answermode=auto;transport=udp;audio_player=alsa,plughw:0,0,3;audio_source=alsa,plughw:0,0,3;ausrc_srate=8000
<sip:flight_deck@0.0.0.0:5060>;regint=1;answermode=auto;transport=udp;audio_player=alsa,plughw:0,0,4;audio_source=alsa,plughw:0,0,4;ausrc_srate=8000
<sip:hanger@0.0.0.0:5060>;regint=1;answermode=auto;transport=udp;audio_player=alsa,plughw:0,0,5;audio_source=alsa,plughw:0,0,5;ausrc_srate=8000
<sip:loudhailer@0.0.0.0:5060>;regint=1;answermode=auto;transport=udp;audio_player=alsa,plughw:0,0,6;audio_source=alsa,plughw:0,0,6;ausrc_srate=8000

config:

sip_listen              0.0.0.0:5061

call_max_calls          20
call_accept             yes

module_path             /usr/local/lib64/baresip/modules
module                  alsa.so
module                  g711.so
module                  account.so
module                  uuid.so
module                  menu.so
audio_codes             pcma

sip_trace               yes
log_level               6

I feel like I'm missing something but I can't seem to find the correct configuration. I see there is a sip_autoanswer, but I haven't trying that out yet (Side note, if I did go that route, what is the correct header to set?)

Any help here would be greatly appreciated.

[cspiel1]

• I'll have a look on answermode=auto
• and put more information to the wiki https://github.com/baresip/baresip/wiki/Accounts#sip_autoanswer

[cspiel1]

Please provide more details:

• Try with main HEAD. There is a fix for "delayed answer": 1e0deeb
• Should the accounts really register at a SIP registrar? If not you should set regint=0. If yes then 1 second is very low. This produces a lot of re-REGISTER traffic. Set it to e.g. regint=120 very low or regint=3600 more typical.
• Which of the accounts are called on first/second call?
• Provide the log which shows first and second call! Note, first set regint adequate!
  • Run baresip with baresip -v -s which shows verbose log + SIP sequence

[cspiel1]

Okay, maybe this needs a fix for autoanswer. I'll check this.

I updated SIP auto-answer in the wiki: https://github.com/baresip/baresip/wiki/Accounts#sip_autoanswer

[cspiel1]

PR: #3710

[cody-herbst]

Sorry for the delay. Just got to work in my time lol. Let me set a little more back story to what I'm trying to accomplish with this.

I want baresip to act as a interface to the alsa devices on my machine. So we have an audio card and for every port on the audio card I want to create a baresip user agent to act as a proxy to that alsa subdevice. I'm using asterisk as my pbx and hoping to bridge the calls to allow announcements. Essentially like an intercom system. In a perfect world, when my Asterisk ARI application initializes it makes calls to the correct user agents and those calls are maintained permanently. From there it will simply be a matter of moving the call channels in and out of the correct bridges.

Asterisk does have some console related functionality but from what I can tell its support is very limited and I ran into issues with bridging console channels together in a conference fashion. So I'm hoping baresip can suit my needs here. In the far future I would like to make my own asterisk channel driver that supports hardware interfaces a little better but that will not be a simple effort.

FYI I was intentionally keeping the SIP reg_int very low. My reasoning for this is if asterisk every crashed, and was reset, I want the client to re-register as quickly as possible. I know there are race case conditions I will need to sort out regarding this but that was my thought process. Full disclosure I'm not an expert in SIP by any means lol. I'm just doing my best with what I know to make this work. So if you guys have any thoughts I'd love to hear them.

I'll pull down the PR and try it out later today. Thank you for your help with this!!!!

[cspiel1]

This now is a little bit off topic. Let's concentrate on what baresip should do here.
With this PR: #3710
and config:

call_hold_other_calls     no

multiple incoming calls can be automatically accepted and get established in parallel. Is this what you expect?

[cody-herbst]

Yes. I pulled it down yesterday and multiple calls can are being answered in parallel. The console output only displays the time/audio for the first call I believe but that is a minor thing. Thank you!