Archives : November-2015
When I do a SIP URI call from my softphone, the call is made directly to the destination host (p2p), bypassing the PBX. So I lose the possibility of recording, making statistics, etc …Is there a way to force URI calls through the PBX? I have fo..
I am trying to set up a default outbound endpoint for my Asterisk 13.6.0PBX, and per https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip, I do in pjsip.conf:[global]default_outbound_endpoint=SillyEndpoint…[SillyEndpoint]type=endpointetc.Howev..
HiI have a 3 level nested while-endwhile loop in a macro that when the execution reaches endwhile, it is jumping out to the While at the caller macro.It shouldnt since the are instructions after the endwhile.– Executing [s@macro-call-from-outside:..
Reminder: speakers deadline this Friday, 27 November at 23:59 UTCWe have already received several really exciting talk proposals but there is still time for people to propose talks or encourage friends or colleagues to speak.Many other dev-rooms a..
Im not sure what is going on there but I wanted to mention that Asterisk 1.8 is completely EOL, there will be no further fixes, even security fixes. For new installations you should use Asterisk 13 which is the most recent LTS.https://wiki.asterisk.org/wiki/display/AST/Asteris..
I have a puzzling situation, and would be grateful for any insight.I have a dialplan that forwards an incoming call out to another number via the same SIP trunk as it came in on. e.g.[from-siptrunk]exten => 0123456789,1,NoOp exten => 0123456789,n,Dial(SIP/siptrunk/0987654321)N..
AllAfter a Dial() I get:WARNING[7964][C-000075a8]: app_dial.c:2437 dial_exec_full: Unable to create channel of type SIP (cause 20 – Subscriber absent)if the subscriber is not registered.Is there a way from dialplan to know, *before* Dial(), if a destinat..
I have a somewhat confusing use case.We use a mobile voip app and our users connect to our PBX via a public IP of our firewall which port forwards to asterisk (TLS and SRTP ports). Works fine. Sometimes however, our users are also connected to our ..
Good day Asterisk users, If this is the wrong place to post this, my apologies. However, Im trying to see where I can get a patch for the res_musiconhold.so module. I have an issue where if someone is placed on hold, or is placed in a queue, after ..
Good day Asterisk users, If this is the wrong place to post this, my apologies. However, Im trying to see where I can get a patch for the res_musiconhold.so module. I have an issue where if someone is placed on hold, or is placed in a queue, after ..