Hi everyone,
I have a SIP trunk from asterisk PBX (Asterisk 11.3.0) to another IP PBX.
I have a client extension 2000 registered on asterisk (using Blink) and I’m placing a call over SIP trunk.
I’m able to see the RTPAUDIOQOS statistics in CDR .
-- Executing [8802@default:1] Log("SIP/2000-00000007", "NOTICE, Dialing out from "asterisk_b link" <2000> to 802 through primary") in new stack
[Apr 16 22:06:13] NOTICE[9560][C-00000005]: Ext. 8802:1 @ default: Dialing out from "asterisk_b link" <2000> to 802 through primary
-- Executing [8802@default:2] Dial("SIP/2000-00000007", "SIP/primary/802,60") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/primary/802
-- SIP/primary-00000008 is ringing
-- SIP/primary-00000008 answered SIP/2000-00000007
-- Locally bridging SIP/2000-00000007 and SIP/primary-00000008
-- Executing [h@default:1] Set("SIP/2000-00000007", "CDR(userfield)=ssrc=1510834108;themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000") in new stack
== Spawn extension (default, 8802, 2) exited non-zero on 'SIP/2000-00000007'
Now I’m trying to do something similar by using the autodial (using a .call file) , placing a call from extension 100 to the same destination over the SIP trunk.
The call gets connected but for some reason the CDR statistics from RTPAUDIOQOS are not working .
-- Attempting call on SIP/primary/802 for 100@autodialer:2 (Retry 1)
== Using SIP RTP CoS mark 5
> Channel SIP/primary-0000000a was answered
-- Executing [100@autodialer:2] Wait("SIP/primary-0000000a", "10") in new stack
-- Executing [100@autodialer:3] Playback("SIP/primary-0000000a", "demo-thanks") in new stack
-- <SIP/primary-0000000a> Playing 'demo-thanks.gsm' (language 'en')
-- Executing [100@autodialer:4] Hangup("SIP/primary-0000000a", "") in new stack
== Spawn extension (autodialer, 100, 4) exited non-zero on 'SIP/primary-0000000a'
-- Executing [h@autodialer:1] Set("SIP/primary-0000000a", "CDR(userfield)=") in new stack
The .call file looks like this (and is placed in /var/spool/asterisk/outgoing/
[root@OpenNMS asterisk]# cat testcall1.call
Channel: SIP/primary/802
CallerID: "Autodialer"<100>
Context: autodialer
Extension: 100
Priority: 2
StartRetry: 20942 1 (1366139947)
My extensions.conf file looks like this:
exten => 2000,1,Dial(SIP/2000)
; Calls placed over SIP trunk
exten => _8XXX,1,Log(NOTICE, Dialing out from ${CALLERID(all)} to ${EXTEN:1} through primary)
exten => _8XXX,n,Dial(SIP/primary/${EXTEN:1},60)
exten => _8XXX,n,Playback(hello-world)
;exten => _8XXX,n,NoOp(RTPAUDIOQOS: ${CHANNEL(rtpqos,audio,all)})
;exten => h,1,NoOp(RTPAUDIOQOS: ${CHANNEL(rtpqos,audio,all)})
exten => h,1,Set(CDR(userfield)=${RTPAUDIOQOS})
exten => _8XXX,n,Playtones(congestion)
exten => _8XXX,n,Hangup()
[autodialer]
exten => h,1,Set(CDR(userfield)=${RTPAUDIOQOS})
exten => 100,1,Playback(demo-instruct)
;exten => h,1,NoOp(RTPAUDIOQOS: ${CHANNEL(rtpqos,audio,all)})
exten => 100,n,Wait(10)
exten => 100,n,Playback(demo-thanks)
exten => 100,n,Hangup
Can you guys please tell me or guide me how to fix this ?
(I’m new to asterisk)