The rest lines in extentions.conf are like
exten => 101,1,Dial(SIP/101,60)
and there are some specific call forwarding numbers/rules, access to the voicemail etc.
This file is also not large ~2kbytes.
I updated code under spoiler: add missing section of the extensions.conf please check.
transport= is commented at working sip.conf. I can see some previous sip.conf.bak, which is not in use, with transport=udp, so...
Yes, only in this [100] section. Other clients seems don't have problem with...
Well, pjsip.conf here (without any changes, untouched after installations) contains some trunk related settings, but all of them are commented. No any in the sip.conf.
And as you can see from my settings, chan_sip makes external calls directly to linphone ident ;).
This is not related to early video setting, sorry. The rest parts of the configuration files are related to the other SIP ## processing, voicemails, etc.
@NoFate
PS The configuration above certainly contains only SIP ##100/101 related processing.
What about: SIP/[email protected]?
PPS Why docker, if you rebuild asterisk from sources?
The difference is the chain processing, when the call is coming from the Intercom (SIP #100), which is not required for the calls to SIP #101 from other SIP ##.
I don't remember full details, sorry, I believe, on linphone app there was no video at all without this option.
The reason could be that my asterisk server is sitting behind strong NAT without white IP address.
Probably you can find some explanation here Asterisk sip.conf canreinvite option...
Yes, please check this post under spoiler.
I don't use HA ATM, but I'm interesting to add this functions to my local cctv server if it is possible to run/adopt your add-ons as standalone service.