search for: configuration_res_pjsip

Displaying 5 results from an estimated 5 matches for "configuration_res_pjsip".

2020 Jun 05
2
pjsip subscribecontext support
Hello, I would like to ask about current state of subscribecontext in pjsip. I found out some 6 years old discussion on that without any plans to implement it in the future. I have phones in different contexts. I suspect, when I use its context to subscribe, they will not see phones from the different contexts. Am I right? Marek
2019 Nov 28
2
PJSIP device_state_busy_at, how does this work?
Hi Gang According to: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk12Configuration_res_pjsip-endpoint_device_state_busy_at And endpoint should return busy if this number is reached. We have PBX Trunks registering to the Asterisk. So we want to limit the number of concurrent calls to a PBX and return busy, if more than the configured number of channels a...
2015 Mar 18
1
pjsip: outofcall_message_context
Hello. Is there an analog option "outofcall_message_context" for pjsip? or: how to determine that the "call" is an outbound text message? Dmitriy Serov.
2020 Jun 05
0
pjsip subscribecontext support
...fferent contexts. I suspect, when I use its context > to subscribe, they will not see phones from the different contexts. Am > I right? > I don't recall when the option was implemented but it's present on the endpoint[1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_res_pjsip#Asterisk16Configuration_res_pjsip-endpoint_subscribe_context -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/as...
2023 Jan 31
1
set codec based on B side
...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. I see these options on this link, https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Configuration_res_pjsip codec_prefs_incoming_offer codec_prefs_outgoing_offer codec_prefs_incoming_answer codec_prefs_outgoing_answer but I dont see them on my pjsip.conf file. I only see these tow: incoming_call_offer_pref outgoing_call_offer_pref Do I have to use the 4 of them on each endoint Alice and Bob? Or just...