Not getting inbound calls

[rambutharaju]

Hi.

We have configured VICIBOX 8 (India) on my server and its on behind the sonic firewall and we have hosted pbx server at UK . Both are connected through VPN and same network.

In vicidial,
We have created an Inbound group,Inbound camp,IN trunk and DID route in VICIdial.
and we also created extensions (phones).
When we are dialing my DID number through UK Hosted pbx, the call is ringing but call not hitting my vicidial server.
Could you please anyone suggest me is there anything wrong with my dial plan entry and carrier ?

My server Ip: 10.12.X.X (Local IP) configured behind the sonic firewall.

Please check my carrier :

Carrier Name: 747XXXX
Registration String: register =>74XXX:gXXXX@sip.XXXX.co.uk:5060
Account Entry:
[74XXXX]
username=74XXXX
type=friend
secret=gXXXX
nat=yes
insecure=port,invite
host=sip.XXXXX.co.uk
fromuser=74XXXX
fromdomain=sip.XXXXX.co.uk
dtmfmode=rfc2833
disallow=all
context=default
allow=g729
allow=alaw
allow=ulaw
Protocol: SIP

Globals String: 7XXX = SIP/74XXX

Dialplan Entry:
exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(${7479XX}/${EXTEN:2},,To)
exten => _9XXXXXXXXXX,3,Hangup

exten => _9.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9.,2,Dial(${7479XX}/${EXTEN:2},,tTor)
exten => _9.,3,Hangup

exten => _.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _.,2,Dial(${74XXX}/${EXTEN},,tTor)
exten => _.,3,Hangup

NOTE: If we are calling through UK server to VICIDAIL DID number: 013XXXXXXXX, call not even hitting to my server.
Could you please suggest where I did mistake in settings? Also, guide me the settings if any changes.

[ambiorixg12]

Make sure the INVITE is reaching your server, carrier can tell you that verifying on their end a 408 Request Timeout is not received, also you can enable sip set debug on and verify your self, in case INVITE doesnt reach your server verify the firewall rules inside and outside vici

[rambutharaju]

Hi Ambiorixg12
Thanks for you promt reply.
As per your suggetion i did SIP set debug on.
I am posting the output here.could you please look into it.
I hope it helps you to correct me.

adroitdial*CLI> sip set debug on
SIP Debugging re-enabled
[May 28 16:17:44] NOTICE[18046]: chan_sip.c:15753 sip_reregister: -- Re-registration for 74xxxx@sip.xxxx.couk
[May 28 16:17:44] REGISTER 12 headers, 0 lines
[May 28 16:17:44] Reliably Transmitting (NAT) toXXXX:5060:
[May 28 16:17:44] REGISTER sip:sip.XXXX.couk SIP/2.0
[May 28 16:17:44] Via: SIP/2.0/UDP XXXX:5060;branch=z9hG4bK2a248b60;rport
[May 28 16:17:44] Max-Forwards: 70
[May 28 16:17:44] From: <sip:74xxxx@sip.xxxx.couk>;tag=as0ec80185
[May 28 16:17:44] To: <sip:74xxxx@sip.xxxx.couk>
[May 28 16:17:44] Call-ID: 71b1b6914eb9b3b10c489e07131b9822@10.12.9.198
[May 28 16:17:44] CSeq: 1424 REGISTER
[May 28 16:17:44] Supported: replaces, timer
[May 28 16:17:44] User-Agent: Asterisk PBX 13.21.1-vici
[May 28 16:17:44] Authorization: Digest username="74XXXX", realm="sip.XXXX.couk", algorithm=MD5, uri="sip:sip.XXXX.couk", nonce="5ced748f000072b83f057f88152d534be211fb20a3947e9e", response="24d36c10054dc7f0aa30b2372ae55de4"
[May 28 16:17:44] Expires: 120
[May 28 16:17:44] Contact: <sip:s@XXXX:5060>
[May 28 16:17:44] Content-Length: 0
[May 28 16:17:44]
[May 28 16:17:44]
[May 28 16:17:44] ---
[May 28 16:17:44]
[May 28 16:17:44] <--- SIP read from UDP:XXXXX:5060 --->
[May 28 16:17:44] SIP/2.0 100 Trying
[May 28 16:17:44] Via: SIP/2.0/UDP XXXX:5060;branch=z9hG4bK2a248b60;rport=31820
[May 28 16:17:44] From: <sip:74xxxx@sip.xxxx.couk>;tag=as0ec80185
[May 28 16:17:44] To: <sip:74xxxx@sip.xxxx.couk>
[May 28 16:17:44] Call-ID: 71b1b6914eb9b3b10c489e07131b9822@10.12.9.198
[May 28 16:17:44] CSeq: 1424 REGISTER
[May 28 16:17:44] Server: OpenSIPS (1.5.3-notls (x86_64/linux))
[May 28 16:17:44] Content-Length: 0
[May 28 16:17:44]
[May 28 16:17:44] <------------->
[May 28 16:17:44] --- (8 headers 0 lines) ---
[May 28 16:17:44]
[May 28 16:17:44] <--- SIP read from UDP:XXXXX:5060 --->
[May 28 16:17:44] SIP/2.0 401 Unauthorized
[May 28 16:17:44] Via: SIP/2.0/UDP XXXX:5060;branch=z9hG4bK2a248b60;rport=31820
[May 28 16:17:44] From: <sip:74xxxx@sip.xxxx.couk>;tag=as0ec80185
[May 28 16:17:44] To: <sip:74xxxx@sip.xxxx.couk>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.5072
[May 28 16:17:44] Call-ID: 71b1b6914eb9b3b10c489e07131b9822@10.12.9.198
[May 28 16:17:44] CSeq: 1424 REGISTER
[May 28 16:17:44] WWW-Authenticate: Digest realm="sip.XXXX.couk", nonce="5ced74f90000aac5e5d199b7f89385e325404b14be78f7bc", stale=true
[May 28 16:17:44] Server: OpenSIPS (1.5.3-notls (x86_64/linux))
[May 28 16:17:44] Content-Length: 0
[May 28 16:17:44]
[May 28 16:17:44] <------------->
[May 28 16:17:44] --- (9 headers 0 lines) ---
[May 28 16:17:44] Responding to challenge, registration to domain/host name sip.XXXX.couk
[May 28 16:17:44] REGISTER 12 headers, 0 lines
[May 28 16:17:44] Reliably Transmitting (NAT) toXXXX:5060:
[May 28 16:17:44] REGISTER sip:sip.XXXX.couk SIP/2.0
[May 28 16:17:44] Via: SIP/2.0/UDP XXXX:5060;branch=z9hG4bK2bef8a44;rport
[May 28 16:17:44] Max-Forwards: 70
[May 28 16:17:44] From: <sip:74xxxx@sip.xxxx.couk>;tag=as0ec80185
[May 28 16:17:44] To: <sip:74xxxx@sip.xxxx.couk>
[May 28 16:17:44] Call-ID: 71b1b6914eb9b3b10c489e07131b9822@10.12.9.198
[May 28 16:17:44] CSeq: 1425 REGISTER
[May 28 16:17:44] Supported: replaces, timer
[May 28 16:17:44] User-Agent: Asterisk PBX 13.21.1-vici
[May 28 16:17:44] Authorization: Digest username="74XXXX", realm="sip.XXXX.couk", algorithm=MD5, uri="sip:sip.XXXX.couk", nonce="5ced74f90000aac5e5d199b7f89385e325404b14be78f7bc", response="9c86f6a43b935ae056ab8d7bd6abc7ea"
[May 28 16:17:44] Expires: 120
[May 28 16:17:44] Contact: <sip:s@XXXX:5060>
[May 28 16:17:44] Content-Length: 0
[May 28 16:17:44]
[May 28 16:17:44]
[May 28 16:17:44] ---
[May 28 16:17:44]
[May 28 16:17:44] <--- SIP read from UDP:XXXXX:5060 --->
[May 28 16:17:44] SIP/2.0 100 Trying
[May 28 16:17:44] Via: SIP/2.0/UDP XXXX:5060;branch=z9hG4bK2bef8a44;rport=31820
[May 28 16:17:44] From: <sip:74xxxx@sip.xxxx.couk>;tag=as0ec80185
[May 28 16:17:44] To: <sip:74xxxx@sip.xxxx.couk>
[May 28 16:17:44] Call-ID: 71b1b6914eb9b3b10c489e07131b9822@10.12.9.198
[May 28 16:17:44] CSeq: 1425 REGISTER
[May 28 16:17:44] Server: OpenSIPS (1.5.3-notls (x86_64/linux))
[May 28 16:17:44] Content-Length: 0
[May 28 16:17:44]
[May 28 16:17:44] <------------->
[May 28 16:17:44] --- (8 headers 0 lines) ---
[May 28 16:17:44]
[May 28 16:17:44] <--- SIP read from UDP:XXXXX:5060 --->
[May 28 16:17:44] SIP/2.0 200 OK
[May 28 16:17:44] Via: SIP/2.0/UDP XXXX:5060;branch=z9hG4bK2bef8a44;rport=31820
[May 28 16:17:44] From: <sip:74xxxx@sip.xxxx.couk>;tag=as0ec80185
[May 28 16:17:44] To: <sip:74xxxx@sip.xxxx.couk>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.1ca2
[May 28 16:17:44] Call-ID: 71b1b6914eb9b3b10c489e07131b9822@10.12.9.198
[May 28 16:17:44] CSeq: 1425 REGISTER
[May 28 16:17:44] Contact: <sip:s@XXXX:5060>;expires=120
[May 28 16:17:44] Server: OpenSIPS (1.5.3-notls (x86_64/linux))
[May 28 16:17:44] Content-Length: 0
[May 28 16:17:44]
[May 28 16:17:44] <------------->
[May 28 16:17:44] --- (9 headers 0 lines) ---
[May 28 16:17:44] NOTICE[18046]: chan_sip.c:24591 handle_response_register: Outbound Registration: Expiry for sip.XXXX.couk is 120 sec (Scheduling reregistration in 105 s)
[May 28 16:17:44] Really destroying SIP dialog '71b1b6914eb9b3b10c489e07131b9822@10.12.9.198' Method: REGISTER
[May 28 16:18:13] Reliably Transmitting (NAT) toXXXX:5060:
[May 28 16:18:13] OPTIONS sip:sip.XXXX.couk SIP/2.0
[May 28 16:18:13] Via: SIP/2.0/UDP XXXX:5060;branch=z9hG4bK74b868e0;rport
[May 28 16:18:13] Max-Forwards: 70
[May 28 16:18:13] From: "asterisk" <sip:74XXXX@XXXX>;tag=as581dca74
[May 28 16:18:13] To: <sip:sip.XXXX.couk>
[May 28 16:18:13] Contact: <sip:74XXXX@XXXX:5060>
[May 28 16:18:13] Call-ID: 2b1691776eb02f0e266474d71defda2b@XXXX:5060
[May 28 16:18:13] CSeq: 102 OPTIONS
[May 28 16:18:13] User-Agent: Asterisk PBX 13.21.1-vici
[May 28 16:18:13] Date: Tue, 28 May 2019 10:48:13 GMT
[May 28 16:18:13] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[May 28 16:18:13] Supported: replaces, timer
[May 28 16:18:13] Content-Length: 0
[May 28 16:18:13]
[May 28 16:18:13]
[May 28 16:18:13] ---
[May 28 16:18:13]
[May 28 16:18:13] <--- SIP read from UDP:XXXXX:5060 --->
[May 28 16:18:13] SIP/2.0 200 OK
[May 28 16:18:13] Via: SIP/2.0/UDP XXXX:5060;branch=z9hG4bK74b868e0;rport=31820
[May 28 16:18:13] From: "asterisk" <sip:74XXXX@XXXX>;tag=as581dca74
[May 28 16:18:13] To: <sip:sip.XXXX.couk>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.d12a
[May 28 16:18:13] Call-ID: 2b1691776eb02f0e266474d71defda2b@XXXX:5060
[May 28 16:18:13] CSeq: 102 OPTIONS
[May 28 16:18:13] Server: OpenSIPS (1.5.3-notls (x86_64/linux))
[May 28 16:18:13] Content-Length: 0
[May 28 16:18:13]
[May 28 16:18:13] <------------->
[May 28 16:18:13] --- (8 headers 0 lines) ---
[May 28 16:18:13] Really destroying SIP dialog '2b1691776eb02f0e266474d71defda2b@XXXX:5060' Method: OPTIONS
[May 28 16:19:13] Reliably Transmitting (NAT) toXXXX:5060:
[May 28 16:19:13] OPTIONS sip:sip.XXXX.couk SIP/2.0
[May 28 16:19:13] Via: SIP/2.0/UDP XXXX:5060;branch=z9hG4bK7470a42a;rport
[May 28 16:19:13] Max-Forwards: 70
[May 28 16:19:13] From: "asterisk" <sip:74XXXX@XXXX>;tag=as6e506769
[May 28 16:19:13] To: <sip:sip.XXXX.couk>
[May 28 16:19:13] Contact: <sip:74XXXX@XXXX:5060>
[May 28 16:19:13] Call-ID: 143ed2c64d4b38f058a645eb70298ccb@XXXX:5060
[May 28 16:19:13] CSeq: 102 OPTIONS
[May 28 16:19:13] User-Agent: Asterisk PBX 13.21.1-vici
[May 28 16:19:13] Date: Tue, 28 May 2019 10:49:13 GMT
[May 28 16:19:13] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[May 28 16:19:13] Supported: replaces, timer
[May 28 16:19:13] Content-Length: 0

[May 28 16:20:13]
[May 28 16:20:13] ---
[May 28 16:20:13]
[May 28 16:20:13] <--- SIP read from UDP:XXXXX:5060 --->
[May 28 16:20:13] SIP/2.0 200 OK
[May 28 16:20:13] Via: SIP/2.0/UDP XXXX:5060;branch=z9hG4bK33966748;rport=7738
[May 28 16:20:13] From: "asterisk" <sip:74XXXX@XXXX>;tag=as01e226de
[May 28 16:20:13] To: <sip:sip.XXXX.couk>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.7837
[May 28 16:20:13] Call-ID: 3c577b5f3383e33218f4f8295aced481@XXXX:5060
[May 28 16:20:13] CSeq: 102 OPTIONS
[May 28 16:20:13] Server: OpenSIPS (1.5.3-notls (x86_64/linux))
[May 28 16:20:13] Content-Length: 0
[May 28 16:20:13]
[May 28 16:20:13] <------------->
[May 28 16:20:13] --- (8 headers 0 lines) ---
[May 28 16:20:13] Really destroying SIP dialog '3c577b5f3383e33218f4f8295aced481@XXXX:5060' Method: OPTIONS

Also Please, find the output of #sip show peers

adroitdial*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
501/501 (Unspecified) D Yes Yes 0 UNKNOWN
74xxxx2/7xxx2 217.x.x.x.x Yes Yes 5060 OK (140 ms)
gs102/gs102 (Unspecified) D Yes Yes 0 UNKNOWN
3 sip peers [Monitored: 1 online, 2 offline Unmonitored: 0 online, 0 offline]

NOTE : we already allowed complete LAN network in firewall.

[ambiorixg12]

I dont see any INVITE request on your logs, verify what SIP response your carrier is getting when trying to send the INVITE to your server

[williamconley]

Both are connected through VPN and same network.
...
Registration String: register =>74XXX:gXXXX@sip.XXXX.co.uk:5060

If they are on the same network (due to VPN), then they should be using local IP addresses to communicate with each other. not "sip.XXX.co.uk" which will attempt to reach outside the VPN to reach the server and fail due to firewall constraints.