incoming call reaching asterisk but no calls for Agent

[ali.rehan]

I am using VICIbox server is ViciBox 8.1.2 Installed on HYPER-V

OpenSuSE Leap v.42.3 64-bit
Kernel v.4.4.155
Asterisk v.13.21.1-vici
DAHDI v.2.11.1
LibPRI v.1.6.0
Amfletec VoiceSync v.1.3.8
OpenR2 v.1.3.3 for MFC/R2 support
ViciDial SVN v.2.14-689a build 180922-0958 revision 3035

The Problem :
Outbound call is working fine but Incoming calls are not recieved by logged in agents although they are reaching asterisk i have purchased the manual and create inbound trunk , Ingroup,compaign and point did to ingroup .






Cli Results:
localhost*CLI>
[Jan 22 09:12:47] == Using SIP RTP CoS mark 5
[Jan 22 09:12:47] > 0x7fd78c027100 -- Strict RTP learning after remote address set to: 204.11.192.169:59846
[Jan 22 09:12:47] -- Executing [17778126342100@trunkinbound:1] Ringing("SIP/122.129.77.114-00000005", "") in new stack
[Jan 22 09:12:47] -- Executing [17778126342100@trunkinbound:2] Wait("SIP/122.129.77.114-00000005", "1") in new stack
[Jan 22 09:12:48] -- Executing [17778126342100@trunkinbound:3] Answer("SIP/122.129.77.114-00000005", "") in new stack
[Jan 22 09:12:49] -- Executing [17778126342100@trunkinbound:4] AGI("SIP/122.129.77.114-00000005", "agi-DID_route.agi") in new stack
[Jan 22 09:12:49] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Jan 22 09:12:49] -- <SIP/122.129.77.114-00000005>AGI Script agi-DID_route.agi completed, returning 0
[Jan 22 09:12:49] -- Executing [17778126342100@trunkinbound:5] Hangup("SIP/122.129.77.114-00000005", "") in new stack
[Jan 22 09:12:49] == Spawn extension (trunkinbound, 17778126342100, 5) exited non-zero on 'SIP/122.129.77.114-00000005'

Trunkinbound
exten => _1x.,1,Ringing
exten => _1x.,2,Wait(1)
exten => _1x.,3,Answer
exten => _1x.,4,AGI(agi-DID_route.agi)
exten => _1x.,5,Hangup

My Inbound Trunk Configuration :
[Callcentric]
type=peer
disallow=all
allow=alaw
allow=ulaw
type=friend
username=17778126342100
secret=xxx
host=callcentric.com
dtmfmode=rfc2833
context=trunkinbound
insecure=very
nat=force_rport,comedia
fromdomain=callcentric.com
defaultuser=17778126342100
fromuser=17778126342100
disallowed_methods=UPDATE
directmedia=no
videosupport=no
canreinvite=no
[callcentric1](callcentric)
host=alpha1.callcentric.com

[callcentric2](callcentric)
host=alpha2.callcentric.com

[callcentric3](callcentric)
host=alpha3.callcentric.com

[callcentric4](callcentric)
host=alpha4.callcentric.com

[callcentric5](callcentric)
host=alpha5.callcentric.com

[callcentric6](callcentric)
host=alpha6.callcentric.com

[callcentric7](callcentric)
host=alpha7.callcentric.com

[callcentric8](callcentric)
host=alpha8.callcentric.com

[callcentric9](callcentric)
host=alpha9.callcentric.com

[callcentric10](callcentric)
host=alpha10.callcentric.com

[callcentric11](callcentric)
host=alpha11.callcentric.com

[callcentric12](callcentric)
host=alpha12.callcentric.com

[callcentric13](callcentric)
host=alpha13.callcentric.com

[callcentric14](callcentric)
host=alpha14.callcentric.com

[callcentric15](callcentric)
host=alpha15.callcentric.com

[callcentric16](callcentric)
host=alpha16.callcentric.com

[callcentric17](callcentric)
host=alpha17.callcentric.com

[callcentric18](callcentric)
host=alpha18.callcentric.com

[callcentric19](callcentric)
host=alpha19.callcentric.com

[callcentric20](callcentric)
host=alpha20.callcentric.com

[callcentricA](callcentric)
host=doll3.callcentric.com

[callcentricB](callcentric)
host=doll4.callcentric.com

[callcentricC](callcentric)
host=doll5.callcentric.com


Debug SIP:
<------------>
[Jan 22 09:28:08] -- Executing [17778126342100@trunkinbound:1] Ringing("SIP/66.193.176.35-00000008", "") in new stack
[Jan 22 09:28:08]
[Jan 22 09:28:08] <--- Transmitting (NAT) to 204.11.192.171:5080 --->
[Jan 22 09:28:08] SIP/2.0 180 Ringing
[Jan 22 09:28:08] Via: SIP/2.0/UDP 204.11.192.171:5080;branch=z9hG4bK-600cdb1c3b0aebb4a6808d63647bdb89;received=204.11.192.171;rport=5080
[Jan 22 09:28:08] From: <sip:9547937099102@66.193.176.35>;tag=3757119845-825643
[Jan 22 09:28:08] To: <sip:18772150306@ss.callcentric.com>;tag=as483fd856
[Jan 22 09:28:08] Call-ID: 3011749-3757119845-825596@msw2.telengy.net
[Jan 22 09:28:08] CSeq: 1 INVITE
[Jan 22 09:28:08] Server: Asterisk PBX 13.21.1-vici
[Jan 22 09:28:08] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jan 22 09:28:08] Supported: replaces, timer
[Jan 22 09:28:08] Contact: <sip:17778126342100@122.129.77.114:5060>
[Jan 22 09:28:08] Content-Length: 0
[Jan 22 09:28:08]
[Jan 22 09:28:08]
[Jan 22 09:28:08] <------------>
[Jan 22 09:28:08] -- Executing [17778126342100@trunkinbound:2] Wait("SIP/66.193.176.35-00000008", "1") in new stack
[Jan 22 09:28:08] Retransmitting #3 (NAT) to 209.126.73.134:5060:
[Jan 22 09:28:08] OPTIONS sip:209.126.73.134 SIP/2.0
[Jan 22 09:28:08] Via: SIP/2.0/UDP 122.129.77.114:5060;branch=z9hG4bK5a81a2d4;rport
[Jan 22 09:28:08] Max-Forwards: 70
[Jan 22 09:28:08] From: "asterisk" <sip:asterisk@122.129.77.114>;tag=as5102ec8d
[Jan 22 09:28:08] To: <sip:209.126.73.134>
[Jan 22 09:28:08] Contact: <sip:asterisk@122.129.77.114:5060>
[Jan 22 09:28:08] Call-ID: 452c90a543c9140b4ade67a47eec068d@122.129.77.114:5060
[Jan 22 09:28:08] CSeq: 102 OPTIONS
[Jan 22 09:28:08] User-Agent: Asterisk PBX 13.21.1-vici
[Jan 22 09:28:08] Date: Tue, 22 Jan 2019 04:28:05 GMT
[Jan 22 09:28:08] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jan 22 09:28:08] Supported: replaces, timer
[Jan 22 09:28:08] Content-Length: 0
[Jan 22 09:28:08]


[Jan 22 09:28:09] ---
[Jan 22 09:28:09] -- Executing [17778126342100@trunkinbound:3] Answer("SIP/66.193.176.35-00000008", "") in new stack
[Jan 22 09:28:09] Audio is at 17912
[Jan 22 09:28:09] Adding codec ulaw to SDP
[Jan 22 09:28:09] Adding non-codec 0x1 (telephone-event) to SDP
[Jan 22 09:28:09]
[Jan 22 09:28:09] <--- Reliably Transmitting (NAT) to 204.11.192.171:5080 --->
[Jan 22 09:28:09] SIP/2.0 200 OK
[Jan 22 09:28:09] Via: SIP/2.0/UDP 204.11.192.171:5080;branch=z9hG4bK-600cdb1c3b0aebb4a6808d63647bdb89;received=204.11.192.171;rport=5080
[Jan 22 09:28:09] From: <sip:9547937099102@66.193.176.35>;tag=3757119845-825643
[Jan 22 09:28:09] To: <sip:18772150306@ss.callcentric.com>;tag=as483fd856
[Jan 22 09:28:09] Call-ID: 3011749-3757119845-825596@msw2.telengy.net
[Jan 22 09:28:09] CSeq: 1 INVITE
[Jan 22 09:28:09] Server: Asterisk PBX 13.21.1-vici
[Jan 22 09:28:09] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Jan 22 09:28:09] Supported: replaces, timer
[Jan 22 09:28:09] Contact: <sip:17778126342100@122.129.77.114:5060>
[Jan 22 09:28:09] Content-Type: application/sdp
[Jan 22 09:28:09] Content-Length: 259


[Jan 22 09:28:10] ---
[Jan 22 09:28:10] -- Executing [17778126342100@trunkinbound:4] AGI("SIP/66.193.176.35-00000008", "agi-DID_route.agi") in new stack
[Jan 22 09:28:10] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Jan 22 09:28:10] -- <SIP/66.193.176.35-00000008>AGI Script agi-DID_route.agi completed, returning 0
[Jan 22 09:28:10] -- Executing [17778126342100@trunkinbound:5] Hangup("SIP/66.193.176.35-00000008", "") in new stack
[Jan 22 09:28:10] == Spawn extension (trunkinbound, 17778126342100, 5) exited non-zero on 'SIP/66.193.176.35-00000008'
[Jan 22 09:28:10] Scheduling destruction of SIP dialog '3011749-3757119845-825596@msw2.telengy.net' in 32000 ms (Method: ACK)
[Jan 22 09:28:10] Reliably Transmitting (NAT) to 204.11.192.171:5080:
[Jan 22 09:28:10] BYE sip:6c16464ab5d765a0955bc1fa9253cefe@204.11.192.171:5080;transport=udp SIP/2.0
[Jan 22 09:28:10] Via: SIP/2.0/UDP 122.129.77.114:5060;branch=z9hG4bK407cb799;rport
[Jan 22 09:28:10] Max-Forwards: 70
[Jan 22 09:28:10] From: <sip:18772150306@ss.callcentric.com>;tag=as483fd856
[Jan 22 09:28:10] To: <sip:9547937099102@66.193.176.35>;tag=3757119845-825643
[Jan 22 09:28:10] Call-ID: 3011749-3757119845-825596@msw2.telengy.net
[Jan 22 09:28:10] CSeq: 102 BYE
[Jan 22 09:28:10] User-Agent: Asterisk PBX 13.21.1-vici
[Jan 22 09:28:10] X-Asterisk-HangupCause: Normal Clearing
[Jan 22 09:28:10] X-Asterisk-HangupCauseCode: 16
[Jan 22 09:28:10] Content-Length: 0

Kindly Help

[williamconley]

you edited the .conf file for "trunkinbound". undo your changes. this applies to any other .conf files. Do not edit the conf files manually. (Sole exception: externip value in sip.conf)

next up: what do you have configured for the inbound DID that arrived? If the DID is configured to route to an ingroup, but you haven't specified WHICH ingroup, this is what happens. Consider routing it to "EXTENSION" and in the extension field put "9998811112" which will play a message and hang up. Most importantly, that message playing successfully will provide proof of sound outbound at least. Or 9994444112 which is a more interesting message.

Code: Select all

Retransmitting #3 (NAT) to 209.126.73.134:5060:

If this is related to the call, you have a firewall problem. If not, turn off the carrier entry for this carrier so it won't pop up during your debug checks and fix that sucker later.

[ali.rehan]

Hi ,
Thank you very much for quick response much appreciated . Actually i have changed a lot while doing hit and trial is there a way get extension.conf and sip.conf to their default settings.

[ali.rehan]

I have rolled back every thing and reinstalled all components again now the problem is i cannot login into agents screen " sorry login/ password are not active in this system "
i have tried creating many users and the result is same ....my sip phone registers with no problems .
kindly help

[williamconley]

I have rolled back every thing and reinstalled all components again now the problem is i cannot login into agents screen " sorry login/ password are not active in this system "
i have tried creating many users and the result is same ....my sip phone registers with no problems .
kindly help

mysql asterisk -e "select user,pass,active from vicidial_users where user_level='9'"

If root mysql user has a password, add -p:
mysql asterisk -e -p "select user,pass,active from vicidial_users where user_level='9'"

IF this is a fresh system that's never worked completely: Seriously consider a fresh install. NEVER fear the reinstall. It's the best reset button you'll ever own. lol. (OK: If this is virtual and you went back to a save point, that's pretty cool, too ... but you didn't learn nuthin' by doin' it!) The more often you start over, the better your understanding of the full system and the less likely you are to panic if (for instance) your HD dies. Oh: Get backups working as soon as you can. There's a built-in script for it that will even push the backup set OFF the server (ie: in case your HD dies, you can rebuild the server with all your cool stuff right back where it was when the HD died! lol)

[ali.rehan]

After doing all roll up this where iam getting now .

n 22 23:54:08] --- (9 headers 0 lines) ---
[Jan 22 23:54:09] -- Executing [9998811112@default:3] Playback("SIP/66.193.176.35-00000002", "ss-noservice") in new stack
[Jan 22 23:54:09] -- <SIP/66.193.176.35-00000002> Playing 'ss-noservice.gsm' (language 'en')
[Jan 22 23:54:09] > 0x7f5ae0018ee0 -- Strict RTP switching to RTP target address 204.11.192.170:59654 as source
[Jan 22 23:54:10] > 0x7f5ae0018ee0 -- Strict RTP learning complete - Locking on source address 204.11.192.170:59654
[Jan 22 23:54:13] -- Executing [9998811112@default:4] Playback("SIP/66.193.176.35-00000002", "vm-goodbye") in new stack
[Jan 22 23:54:14] -- <SIP/66.193.176.35-00000002> Playing 'vm-goodbye.gsm' (language 'en')
[Jan 22 23:54:14] -- Executing [9998811112@default:5] Hangup("SIP/66.193.176.35-00000002", "") in new stack
[Jan 22 23:54:14] == Spawn extension (default, 9998811112, 5) exited non-zero on 'SIP/66.193.176.35-00000002'
[Jan 22 23:54:14] WARNING[5930][C-00000006]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Jan 22 23:54:14] -- Executing [h@default:1] AGI("SIP/66.193.176.35-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Jan 22 23:54:15] -- <SIP/66.193.176.35-00000002>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
[Jan 22 23:54:15] Scheduling destruction of SIP dialog '3627751-3757207803-181086@msw2.telengy.net' in 32000 ms (Method: ACK)
[Jan 22 23:54:15] Reliably Transmitting (NAT) to 204.11.192.170:5080:
[Jan 22 23:54:15] BYE sip:b8436a69a3a57beb92971fd798278a9b@204.11.192.170:5080;transport=udp SIP/2.0
[Jan 22 23:54:15] Via: SIP/2.0/UDP 122.129.77.114:5060;branch=z9hG4bK3bca691b;rport
[Jan 22 23:54:15] Max-Forwards: 70
[Jan 22 23:54:15] From: <sip:18772150306@ss.callcentric.com>;tag=as607db14d
[Jan 22 23:54:15] To: <sip:9547937099102@66.193.176.35>;tag=3757207803-181126
[Jan 22 23:54:15] Call-ID: 3627751-3757207803-181086@msw2.telengy.net
[Jan 22 23:54:15] CSeq: 102 BYE
[Jan 22 23:54:15] User-Agent: Asterisk PBX 13.21.1-vici
[Jan 22 23:54:15] X-Asterisk-HangupCause: Normal Clearing
[Jan 22 23:54:15] X-Asterisk-HangupCauseCode: 16
[Jan 22 23:54:15] Content-Length: 0