Actually I am having no voice issue on some inbound calls. The DID provider insists that its either my issue or it was a spam call
please ignore the following as i posted it earlier and was going in wrong direction
What could possibly the reason for
– Timeout on SIP/xxx.xxx.xxx.xxx-000039cb, continuing…
I received a call which did not had any voice. When I saw the logs I found it was timed-out and then executed the lines in extensions.conf
complete logs for that call:
== Using SIP RTP CoS mark 5
– Executing [19092458367@inbound:1] Answer(“SIP/xxx.xxx.xxx.xxx-000039cb”, “”) in new stack
– Executing [19092458367@inbound:2] Wait(“SIP/xxx.xxx.xxx.xxx-000039cb”, “2”) in new stack
– Executing [19092458367@inbound:3] BackGround(“SIP/xxx.xxx.xxx.xxx-000039cb”, “welcom33”) in new stack
– <SIP/xxx.xxx.xxx.xxx-000039cb> Playing ‘welcom33.slin’ (language ‘en’)
– Executing [19092458367@inbound:4] WaitExten(“SIP/xxx.xxx.xxx.xxx-000039cb”, “3”) in new stack
– Timeout on SIP/xxx.xxx.xxx.xxx-000039cb, continuing…
– Executing [19092458367@inbound:5] Macro(“SIP/xxx.xxx.xxx.xxx-000039cb”, “callrin,51”) in new stack
– Executing [s@macro-callrin:1] Set(“SIP/xxx.xxx.xxx.xxx-000039cb”, “CALLERID(name)=51–19496973777”) in new stack
– Executing [s@macro-callrin:2] Set(“SIP/xxx.xxx.xxx.xxx-000039cb”, “CDR(accountcode)=INBOUND”) in new stack
– Executing [s@macro-callrin:3] Set(“SIP/xxx.xxx.xxx.xxx-000039cb”, “date=2016-09-08”) in new stack
– Executing [s@macro-callrin:4] Set(“SIP/xxx.xxx.xxx.xxx-000039cb”, “time=15-26-21”) in new stack
– Executing [s@macro-callrin:5] Set(“SIP/xxx.xxx.xxx.xxx-000039cb”, “MONITOR_FILENAME=/var/spool/asterisk/monitor/2016-09-08/INBOUND/51–19496973777—15-26-21”) in new stack
– Executing [s@macro-callrin:6] Monitor(“SIP/xxx.xxx.xxx.xxx-000039cb”, “wav,/var/spool/asterisk/monitor/2016-09-08/INBOUND/51–19496973777—15-26-21,m”) in new stack
– Executing [19092458367@inbound:6] Dial(“SIP/xxx.xxx.xxx.xxx-000039cb”, “SIP/102&SIP/104&SIP/100&SIP/103&SIP/106,30,Tt”) in new stack
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
[2016-09-08 15:26:21] WARNING[29980][C-0006c97b]: app_dial.c:2437 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
– Called SIP/1002
– Called SIP/1004
– Called SIP/1000
– Called SIP/1003
– SIP/1004-000039cd is ringing
– SIP/1003-000039cf is ringing
– SIP/1004-000039cd is ringing
– SIP/1002-000039cc is ringing
– SIP/1003-000039cf is ringing
– SIP/1000-000039ce is ringing
– SIP/1002-000039cc answered SIP/xxx.xxx.xxx.xxx-000039cb
I am using Asterisk server with ubuntu