We are having issues with PJSIP and the latest version of Asterisk.When we do the PJSIP list endpoints again it shows n..
Kristian Kielhofner wrote a couple of years ago walking you through his experiences with the SIP..
Is there a way to set a given PJSIP transport to use a given interface and doing so, brings VRF-awareness to..
How can I subscribe over AMI and what response should I look for ifI need to make a script that will SUBSCRIBE to the status of certain SIP..
Finally, I figured out how our SBC does matches invites to registrations with the Contact header, but now I face another issue: how do I set the contact header of an invite different to the From header?INVITE sip:called-id@URI SIP/2.0Via: SIP/2.0/..
How to play the hold music on a channel in a ConfBridge when all the other channels ar..
How to centralize on a single Asterisk instance several Asterisk Installations using PJSIP an..
i know about architecture limits of app_queuehttps://issues.asterisk.org/jira/browse/ASTERISK-25806what CPUs are you actually using for asterisk + app_queue ? (my actual scenario 90simult calls, 50agents, call recording to SSD (mi..
everybody,I am seeing a strange problem on Asterisk 1.8 with dnsmgr.The number of entries in DNS Manager seems to be growing steadily and all are pointing to the sama host – a SIP trunk to a local provider, which uses SRV lookup.So, when DNS mana..
My setup using 11.25.1 was working. When I installed 11.25.2 I now getsort of working.I am using NAT in the setup. When I have an internal phone and call out Iget audio both ways. But when I call IN my phone rings but I have no audio.Is there a new sett..