Archives : January-2022
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I am running 18.8.0 -videosupport is enabled. I get video calls no problem.However when I make a call file to a soft phone and include:Codecs: ulaw,h264in the call file…sip show channels – shows:4013c15f1f4cdff(ulaw|h264)No Tx: ACKso clearly the cal..
I am trying to run this command:exten => _4XX,n,System(/usr/bin/rm /tmp/test.incoming.txt)From the log: Executing [402@smvoice-sip:7] System(SIP/103-00000018, /usr/bin/rm/tmp/test.incoming.txt) in new stackIs rm not an allowed command – the above f..
i have 2 queues – queue1 – queue2 1 agent is in both queues queue strategy is rrmemory i have 2 calls waiting call from 12:00 in queue1 from number 777 call from 12:05 in queue2 from number 666 at 12:10 agent is free for next call i have problem in t..
Hi.I am using asterisk 18.3 and freepbx.How can both sip and pjsip be listening at port 5060 at the same time, for instance I get: [2022-01-08 17:08:59] SECURITY[244351] res_security_log.c: SecurityEvent=FailedACL,EventTV=2022-01-08T17:08:59.957-0500,Severity=Error,Service=PJSIP,EventVersion=1,AccountID=anonymous,SessionID=2025076022,LocalAddress=IPV4/UDP/166.84.7.53/5060,RemoteAddress=IPV4/UDP/45.134.144.118/5823,ACLName=registrar_attempt_without_configured_a..
I have a hangup handler thats added at the beginning of a call. It logs all the call details. Using the CONTEXT variable I am always going to get the context where the code is being ran and not the last context that the caller is in. Is there any creat..
–000000000000a1a4c505d4adaafa Content-Type: text/plain; charset=UTF-8This includes Jira, the Wiki and Gerrit due to loss of internet access.If youre currently signed in to the community forums, youre OK but new logins wont be accepted. The IRC chann..