Tag : PJSIP
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Ive just discovered PJSIP s support of set_var setting in pjsip.conf. Is this setting also supported in pjsip_wizard.conf ?On a fresh 13.8.2, it doesnt seem but I may have missed somthi..
We currently have a production Asterisk box running 1.8.20.1 which uses MeetMe and chan_sip for conferences. I have been testing the new versions of Asterisk with PJSIP and ConfBridge but have run into an issue which is preventing us from moving forwa..
I am trying to set add a SIP Header to a call before adding it to the Queue.The dial plan sends the call to my macro to perform the work.When I use chan_sip, everything works as expected.When I use PSJIP support, its not adding the SIP header. Look..
In PJSIP configuration, I thought that from_domain parameter in a endpoint permits to have two SIP peers with the same usernames with a different domain.Ive tested at the transport level, I see no changes.Ive also tested with realm parameter in a..
, Im trying to use CHANNEL(aor) and CHANNEL(contact) on PJSIP channel, on asterisk-13.3.2, but they dont return anything. Is this a bug, or did I miss something? Here is my test dialplan: exten => *98,1,Answer same => n,NoOp(Channel=,type= ) same..
I found an issue with how PJSIP handles a typo in the Dial application. If the Channel is mistakenly typed with two slashes (i.e Dial(PJSIP//xxxx…), the Dial applications fails (obviously), but it also kills the server.I put some code in my pbx_con..
I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic configuration works, and I am connected to a SIP trunk using SIP.US, and have set up my inbound calling which works correctly (when I call my PBXDID, the call does come into my ..
I want to have Kamailio in front of one or more Asterisk boxes.I dont think it is necessary for Kamailio and Asterisk to register with one another. Id like for PJSIP to recognise Kamailio by its IP address.I have two boxes, both have public IP address..
I have an Asterisk 13 that only processes app Transfer with PJSIP, to the tune of 60 per second. No voice calls. After like 2 hours, I can no longer get into Asterisk. This command, asterisk -r, fails, and also asterisk -rx core show channels, etc..
Im currently evaluating asterisk 13 (Currently on 11).Were testing the migration from SIP to PJSIP.Is there a way to alias the SIP channeltype to PJSIP when exlusively using pjsip? Matt Hoskins | NPG Corp | Systems Architect816.749.2815 (Internal: e..