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...