Incoming call's from SIP Trunk are not working

maragani

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#1
Hello all,

First let me explain you about my setup, I have installed elastix 1.6 with connecting PSTN Lines through GXW4104 and configured SIP Trunk from VoipVoip. I'm able to make and recevice the calls through GXW4104 without any issues. Now issue is with SIP Trunk,that is when i try to make a call to the voipvoip DID, call is hitting to the elastix but it is not transfering the calls to extension or to IVR.
Below is the cli output i had tried calling to sip trunk:
-- Executing [5551939739@from-sip-external:1] NoOp("SIP/VoIPVoIP-b6c11a08", "Received incoming SIP connection from unknown peer to 5551939739") in new stack
-- Executing [5551939739@from-sip-external:2] Set("SIP/VoIPVoIP-b6c11a08", "DID=5551939739") in new stack
-- Executing [5551939739@from-sip-external:3] Goto("SIP/VoIPVoIP-b6c11a08", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/VoIPVoIP-b6c11a08", "1?from-trunk|5551939739|1") in new stack
-- Goto (from-trunk,5551939739,1)
-- Executing [5551939739@from-trunk:1] NoOp("SIP/VoIPVoIP-b6c11a08", "Catch-All DID Match - Found 5551939739 - You probably want a DID for this.") in new stack
-- Executing [5551939739@from-trunk:2] Goto("SIP/VoIPVoIP-b6c11a08", "ext-did|s|1") in new stack
-- Goto (ext-did,s,1)
-- Executing [s@ext-did:1] Set("SIP/VoIPVoIP-b6c11a08", "__FROM_DID=s") in new stack
-- Executing [s@ext-did:2] Gosub("SIP/VoIPVoIP-b6c11a08", "app-blacklist-check|s|1") in new stack
-- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/VoIPVoIP-b6c11a08", "") in new stack
-- Executing [s@app-blacklist-check:2] GotoIf("SIP/VoIPVoIP-b6c11a08", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/VoIPVoIP-b6c11a08", "") in new stack
-- Executing [s@ext-did:3] ExecIf("SIP/VoIPVoIP-b6c11a08", "1 |Set|CALLERID(name)=2485562133") in new stack
-- Executing [s@ext-did:4] Set("SIP/VoIPVoIP-b6c11a08", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [s@ext-did:5] SetCallerPres("SIP/VoIPVoIP-b6c11a08", "allowed_not_screened") in new stack
-- Executing [s@ext-did:6] Goto("SIP/VoIPVoIP-b6c11a08", "ivr-3|s|1") in new stack
-- Goto (ivr-3,s,1)
-- Executing [s@ivr-3:1] Set("SIP/VoIPVoIP-b6c11a08", "MSG=custom/ivr_in_3CX") in new stack
-- Executing [s@ivr-3:2] Set("SIP/VoIPVoIP-b6c11a08", "LOOPCOUNT=0") in new stack
-- Executing [s@ivr-3:3] Set("SIP/VoIPVoIP-b6c11a08", "__DIR-CONTEXT=default") in new stack
-- Executing [s@ivr-3:4] Set("SIP/VoIPVoIP-b6c11a08", "_IVR_CONTEXT_ivr-3=") in new stack
-- Executing [s@ivr-3:5] Set("SIP/VoIPVoIP-b6c11a08", "_IVR_CONTEXT=ivr-3") in new stack
-- Executing [s@ivr-3:6] GotoIf("SIP/VoIPVoIP-b6c11a08", "0?begin") in new stack
-- Executing [s@ivr-3:7] Answer("SIP/VoIPVoIP-b6c11a08", "") in new stack
-- Executing [s@ivr-3:8] Wait("SIP/VoIPVoIP-b6c11a08", "1") in new stack
-- Executing [s@ivr-3:9] Set("SIP/VoIPVoIP-b6c11a08", "TIMEOUT(digit)=3") in new stack
-- Digit timeout set to 3
-- Executing [s@ivr-3:10] Set("SIP/VoIPVoIP-b6c11a08", "TIMEOUT(response)=10") in new stack
-- Response timeout set to 10
-- Executing [s@ivr-3:11] Set("SIP/VoIPVoIP-b6c11a08", "__IVR_RETVM=") in new stack
-- Executing [s@ivr-3:12] ExecIf("SIP/VoIPVoIP-b6c11a08", "1|Background|custom/ivr_in_3CX") in new stack
-- Executing [s@ivr-3:13] WaitExten("SIP/VoIPVoIP-b6c11a08", "|") in new stack
-- Timeout on SIP/VoIPVoIP-b6c11a08, going to 't'
-- Executing [t@ivr-3:1] DBdel("SIP/VoIPVoIP-b6c11a08", "") in new stack
-- Executing [t@ivr-3:2] Set("SIP/VoIPVoIP-b6c11a08", "__NODEST=") in new stack
-- Executing [t@ivr-3:3] Goto("SIP/VoIPVoIP-b6c11a08", "from-did-direct|1000|1") in new stack
-- Goto (from-did-direct,1000,1)
-- Executing [1000@from-did-direct:1] Macro("SIP/VoIPVoIP-b6c11a08", "exten-vm|novm|1000") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/VoIPVoIP-b6c11a08", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/VoIPVoIP-b6c11a08", "AMPUSER=2485562133") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/VoIPVoIP-b6c11a08", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/VoIPVoIP-b6c11a08", "1|Set|REALCALLERIDNUM=2485562133") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/VoIPVoIP-b6c11a08", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/VoIPVoIP-b6c11a08", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/VoIPVoIP-b6c11a08", "1?report") in new stack
-- Goto (macro-user-callerid,s,10)
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/VoIPVoIP-b6c11a08", "0?continue") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/VoIPVoIP-b6c11a08", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/VoIPVoIP-b6c11a08", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/VoIPVoIP-b6c11a08", "Using CallerID "2485562133" <2485562133>") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/VoIPVoIP-b6c11a08", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/VoIPVoIP-b6c11a08", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/VoIPVoIP-b6c11a08", "EXTTOCALL=1000") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/VoIPVoIP-b6c11a08", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("SIP/VoIPVoIP-b6c11a08", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("SIP/VoIPVoIP-b6c11a08", "RT=""") in new stack
-- Executing [s@macro-exten-vm:8] Macro("SIP/VoIPVoIP-b6c11a08", "record-enable|1000|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/VoIPVoIP-b6c11a08", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/VoIPVoIP-b6c11a08", "recordingcheck|20100611-180557|1276259745.707") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20100611-180557|1276259745.707: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("SIP/VoIPVoIP-b6c11a08", "") in new stack
-- Executing [s@macro-exten-vm:9] Macro("SIP/VoIPVoIP-b6c11a08", "dial||tr|1000") in new stack
-- Executing [s@macro-dial:1] GotoIf("SIP/VoIPVoIP-b6c11a08", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("SIP/VoIPVoIP-b6c11a08", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is '2485562133' number is '2485562133'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 1000 to extension map
-- dialparties.agi: Extension 1000 cf is disabled
-- dialparties.agi: Extension 1000 do not disturb is disabled
dialparties.agi: ExtensionState: 0
dialparties.agi: Extension 1000 has ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 1000
-- dialparties.agi: dbset CALLTRACE/1000 to 2485562133
-- dialparties.agi: Filtered ARG3: 1000
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] Dial("SIP/VoIPVoIP-b6c11a08", "SIP/1000||tr") in new stack
-- Couldn't call 1000
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [s@macro-dial:8] Set("SIP/VoIPVoIP-b6c11a08", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-dial:9] GosubIf("SIP/VoIPVoIP-b6c11a08", "0?CHANUNAVAIL|1") in new stack
-- Executing [s@macro-exten-vm:10] GotoIf("SIP/VoIPVoIP-b6c11a08", "0?exit|return") in new stack
-- Executing [s@macro-exten-vm:11] Set("SIP/VoIPVoIP-b6c11a08", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:12] GosubIf("SIP/VoIPVoIP-b6c11a08", "0?docfu|1") in new stack
-- Executing [s@macro-exten-vm:13] GosubIf("SIP/VoIPVoIP-b6c11a08", "0?docfb|1") in new stack
-- Executing [s@macro-exten-vm:14] Set("SIP/VoIPVoIP-b6c11a08", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:15] NoOp("SIP/VoIPVoIP-b6c11a08", "Voicemail is novm") in new stack
-- Executing [s@macro-exten-vm:16] GotoIf("SIP/VoIPVoIP-b6c11a08", "1?s-CHANUNAVAIL|1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-exten-vm:1] NoOp("SIP/VoIPVoIP-b6c11a08", "IVR_RETVM: IVR_CONTEXT: ivr-3") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:2] GotoIf("SIP/VoIPVoIP-b6c11a08", "0?exit|1") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:3] PlayTones("SIP/VoIPVoIP-b6c11a08", "congestion") in new stack
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/VoIPVoIP-b6c11a08' in macro 'exten-vm'
== Spawn extension (from-did-direct, 1000, 1) exited non-zero on 'SIP/VoIPVoIP-b6c11a08'
Please let me know if you all need anything more in assisting me.
Many Thanks in Advance.
 

jgutierrez

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#2
From what I can see the, the is arriving into your elastix server, it is going into your IVR, and playingback an audio called ivr_in_3CX then as no one choose any option from the IVR it goes in the t extension and tries to dial extension 1000, but it is not available.

Are you able to hear the audio? Is extension 1000 registered?
If you execute the following command, are you able to hear the audio:

asterisk -rx "originate SIP/503 application playback custom/ivr_in_3CX"
and what about
asterisk -rx "originate SIP/503 application playback demo-congrats"

Where you must replace 503 with another working SIP extension.
If you cant hear your audio, then it doesn't has a valid asterisk audio format
 

maragani

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#3
Thank you for your response.

Yes you are rite when i tried to calling voipvoip DID call is going to IVR but i am unable to hear the IVR and the call is not getting transfer to the extension.

Another issue is that when i change the Inbound route from IVR to an extension i could still see the call reaching to IVR but not to extensions which i have assigned in the inbound route and i am not getting any audio, after few seconds call is getting disconnected.

After excuting the commands I am able to hear the IVR and Demo congrats.
 

maragani

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#4
Hey...

I am able to make and receive from VoipVoiP SIP trunk now.
Below are changes i have made in the tunks setting us per the voip povider asked me to di it and it got worked
disallow=all
allow=ilbc&ulaw&alaw

Thanks you very much for your support
 

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