vicidial.org

Hi Team,

Inbound is not working in my VICI BOX 11,

They are showing the below error.

NOTICE[2849][C-0000001c]: chan_sip.c:26898 handle_request_invite: Call from '7000' (192.168.1.5:5060) to extension 's' rejected because extension not found in context 'trunkinbound'.
-----------
Sip Debug
------------------------
[Feb 26 15:04:23] <--- SIP read from UDP:192.168.1.5:5060 --->
[Feb 26 15:04:23] OPTIONS sip:s@192.168.1.12:5060 SIP/2.0
[Feb 26 15:04:23] Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bKPje7d5f3ff-71ac-4e4a-8f35-c31cd8cb4346
[Feb 26 15:04:23] From: <sip:unknown@192.168.1.5>;tag=40605cb7-c490-413c-aa4d-bb58ff9e8d55
[Feb 26 15:04:23] To: <sip:s@192.168.1.12>
[Feb 26 15:04:23] Contact: <sip:unknown@192.168.1.5:5060>
[Feb 26 15:04:23] Call-ID: d2aaa518-d34c-4092-9317-911bacc46935
[Feb 26 15:04:23] CSeq: 27033 OPTIONS
[Feb 26 15:04:23] Max-Forwards: 70
[Feb 26 15:04:23] User-Agent: Yeastar
[Feb 26 15:04:23] Content-Length: 0
[Feb 26 15:04:23]
[Feb 26 15:04:23] <------------->
[Feb 26 15:04:23] --- (10 headers 0 lines) ---
[Feb 26 15:04:23] Sending to 192.168.1.5:5060 (NAT)
[Feb 26 15:04:23] Looking for s in trunkinbound (domain 192.168.1.12)
[Feb 26 15:04:23]
[Feb 26 15:04:23] <--- Transmitting (NAT) to 192.168.1.5:5060 --->
[Feb 26 15:04:23] SIP/2.0 404 Not Found
[Feb 26 15:04:23] Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bKPje7d5f3ff-71ac-4e4a-8f35-c31cd8cb4346;received=192.168.1.5;rport=5060
[Feb 26 15:04:23] From: <sip:unknown@192.168.1.5>;tag=40605cb7-c490-413c-aa4d-bb58ff9e8d55
[Feb 26 15:04:23] To: <sip:s@192.168.1.12>;tag=as0d07de39
[Feb 26 15:04:23] Call-ID: d2aaa518-d34c-4092-9317-911bacc46935
[Feb 26 15:04:23] CSeq: 27033 OPTIONS
[Feb 26 15:04:23] Server: Asterisk PBX 16.30.0-vici
[Feb 26 15:04:23] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Feb 26 15:04:23] Supported: replaces, timer
[Feb 26 15:04:23] Accept: application/sdp
[Feb 26 15:04:23] Content-Length: 0
[Feb 26 15:04:23]
[Feb 26 15:04:23]
[Feb 26 15:04:23] <------------>
[Feb 26 15:04:23] Scheduling destruction of SIP dialog 'd2aaa518-d34c-4092-9317-911bacc46935' in 32000 ms (Method: OPTIONS)
[Feb 26 15:04:29] NOTICE[3051]: chan_sip.c:15920 sip_reregister: -- Re-registration for 7000@192.168.1.5
[Feb 26 15:04:29] REGISTER 12 headers, 0 lines
[Feb 26 15:04:29] Reliably Transmitting (NAT) to 192.168.1.5:5060:
[Feb 26 15:04:29] REGISTER sip:192.168.1.5 SIP/2.0
[Feb 26 15:04:29] Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK75c5c52b;rport
[Feb 26 15:04:29] Max-Forwards: 70
[Feb 26 15:04:29] From: <sip:7000@192.168.1.5>;tag=as47d7b105
[Feb 26 15:04:29] To: <sip:7000@192.168.1.5>
[Feb 26 15:04:29] Call-ID: 23f329792636da9a696542a60bb7546f@192.168.1.12
[Feb 26 15:04:29] CSeq: 193 REGISTER
[Feb 26 15:04:29] Supported: replaces, timer
[Feb 26 15:04:29] User-Agent: Asterisk PBX 16.30.0-vici
[Feb 26 15:04:29] Authorization: Digest username="7000", realm="YSAsterisk", algorithm=MD5, uri="sip:192.168.1.5", nonce="1708920165/6b579a871728947c96a1366be2ab60b7", response="08837fc4e69090cad02ede86ed8e8b99", opaque="4bcfc073067f2ecf", qop=auth, cnonce="28bdc9a5", nc=00000002
[Feb 26 15:04:29] Expires: 120
[Feb 26 15:04:29] Contact: <sip:s@192.168.1.12:5060>
[Feb 26 15:04:29] Content-Length: 0
------------------------------------------------

[Carrier]
username=Carrier
secret=Carrierpass
fromuser=Carrier
type=peer
host=192.168.1.5
fromdomain=192.168.1.5
port=5060
qualify=yes
insecure=invite,port
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=invite
context=trunkinbound
nat=force_rport,comedia
dtmfmode=rfc2833

DID is looking like that

Hi,

You are using GSM Gateway?

Try setinng up a Globals String to Your Carrier ID

If that doesn't work setup the carrier sip account username in GSM Gateway to just numeric and use the same numeric value as carrier id and try it should work.

Good Luck

Debasish_das wrote:Hi Team,

Inbound is not working in my VICI BOX 11,

They are showing the below error.

NOTICE[2849][C-0000001c]: chan_sip.c:26898 handle_request_invite: Call from '7000' (192.168.1.5:5060) to extension 's' rejected because extension not found in context 'trunkinbound'.
-----------
Sip Debug
------------------------
[Feb 26 15:04:23] <--- SIP read from UDP:192.168.1.5:5060 --->
[Feb 26 15:04:23] OPTIONS sip:s@192.168.1.12:5060 SIP/2.0
[Feb 26 15:04:23] Via: SIP/2.0/UDP 192.168.1.5:5060;rport;branch=z9hG4bKPje7d5f3ff-71ac-4e4a-8f35-c31cd8cb4346
[Feb 26 15:04:23] From: <sip:unknown@192.168.1.5>;tag=40605cb7-c490-413c-aa4d-bb58ff9e8d55
[Feb 26 15:04:23] To: <sip:s@192.168.1.12>
[Feb 26 15:04:23] Contact: <sip:unknown@192.168.1.5:5060>
[Feb 26 15:04:23] Call-ID: d2aaa518-d34c-4092-9317-911bacc46935
[Feb 26 15:04:23] CSeq: 27033 OPTIONS
[Feb 26 15:04:23] Max-Forwards: 70
[Feb 26 15:04:23] User-Agent: Yeastar
[Feb 26 15:04:23] Content-Length: 0
[Feb 26 15:04:23]
[Feb 26 15:04:23] <------------->
[Feb 26 15:04:23] --- (10 headers 0 lines) ---
[Feb 26 15:04:23] Sending to 192.168.1.5:5060 (NAT)
[Feb 26 15:04:23] Looking for s in trunkinbound (domain 192.168.1.12)
[Feb 26 15:04:23]
[Feb 26 15:04:23] <--- Transmitting (NAT) to 192.168.1.5:5060 --->
[Feb 26 15:04:23] SIP/2.0 404 Not Found
[Feb 26 15:04:23] Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bKPje7d5f3ff-71ac-4e4a-8f35-c31cd8cb4346;received=192.168.1.5;rport=5060
[Feb 26 15:04:23] From: <sip:unknown@192.168.1.5>;tag=40605cb7-c490-413c-aa4d-bb58ff9e8d55
[Feb 26 15:04:23] To: <sip:s@192.168.1.12>;tag=as0d07de39
[Feb 26 15:04:23] Call-ID: d2aaa518-d34c-4092-9317-911bacc46935
[Feb 26 15:04:23] CSeq: 27033 OPTIONS
[Feb 26 15:04:23] Server: Asterisk PBX 16.30.0-vici
[Feb 26 15:04:23] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Feb 26 15:04:23] Supported: replaces, timer
[Feb 26 15:04:23] Accept: application/sdp
[Feb 26 15:04:23] Content-Length: 0
[Feb 26 15:04:23]
[Feb 26 15:04:23]
[Feb 26 15:04:23] <------------>
[Feb 26 15:04:23] Scheduling destruction of SIP dialog 'd2aaa518-d34c-4092-9317-911bacc46935' in 32000 ms (Method: OPTIONS)
[Feb 26 15:04:29] NOTICE[3051]: chan_sip.c:15920 sip_reregister: -- Re-registration for 7000@192.168.1.5
[Feb 26 15:04:29] REGISTER 12 headers, 0 lines
[Feb 26 15:04:29] Reliably Transmitting (NAT) to 192.168.1.5:5060:
[Feb 26 15:04:29] REGISTER sip:192.168.1.5 SIP/2.0
[Feb 26 15:04:29] Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK75c5c52b;rport
[Feb 26 15:04:29] Max-Forwards: 70
[Feb 26 15:04:29] From: <sip:7000@192.168.1.5>;tag=as47d7b105
[Feb 26 15:04:29] To: <sip:7000@192.168.1.5>
[Feb 26 15:04:29] Call-ID: 23f329792636da9a696542a60bb7546f@192.168.1.12
[Feb 26 15:04:29] CSeq: 193 REGISTER
[Feb 26 15:04:29] Supported: replaces, timer
[Feb 26 15:04:29] User-Agent: Asterisk PBX 16.30.0-vici
[Feb 26 15:04:29] Authorization: Digest username="7000", realm="YSAsterisk", algorithm=MD5, uri="sip:192.168.1.5", nonce="1708920165/6b579a871728947c96a1366be2ab60b7", response="08837fc4e69090cad02ede86ed8e8b99", opaque="4bcfc073067f2ecf", qop=auth, cnonce="28bdc9a5", nc=00000002
[Feb 26 15:04:29] Expires: 120
[Feb 26 15:04:29] Contact: <sip:s@192.168.1.12:5060>
[Feb 26 15:04:29] Content-Length: 0
------------------------------------------------

Your inbound number is not showing in the incoming call, it is coming with "s", so you can fix that from your carrier end and pass the number or you can setup register string with inbound number so it will work properly.

No I'm Using SIP trunks,

However I solved it, Thanks Mate For your help.