8007490000- not working from STC SIP Trunk in saudi Arabia

[Eldho E Kuriakose]

Please find the attached traces from our gateway and STC side. STC is saying PBX problem

Can anybody from Saudi test this by dialing 8007490000 from their SIP trunk .

[Shadi Shami]

Looking into this one.

[Shadi Shami · Accepted solution]

Tested it for you and got the same exact result, however it's working fine from FXO/ PRI lines.

*Apr 12 14:32:04.359: //528456/956488000000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:8007490000@X.X.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK39D023D9
Remote-Party-ID: "Jaffar AlMoosa" <sip:XXXXXXX@X.X.X.X>;party=calling;screen=yes;privacy=off
From: "User Name" <sip:XXXXXXX@X.X.X.X>;tag=A953444-DE8
To: <sip:8007490000@X.X.X.X>
Date: Sun, 12 Apr 2015 14:32:04 GMT
Call-ID: 860A521A-E05711E4-A69CF05C-2FA133A4@X.X.X.X
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2506393600-0000065536-0000003249-3473633290
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M6a
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1428849124
Contact: <sip:XXXXXXX@X.X.X.X:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 271

v=0
o=CiscoSystemsSIP-GW-UserAgent 9860 1174 IN IP4 X.X.X.X
s=SIP Call
c=IN IP4 X.X.X.X
t=0 0
m=audio 21954 RTP/AVP 8 101 19
c=IN IP4 X.X.X.X
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20

*Apr 12 14:32:04.375: //528456/956488000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK39D023D9
Call-ID: 860A521A-E05711E4-A69CF05C-2FA133A4@X.X.X.X
From: "Jaffar AlMoosa"<sip:XXXXXXX@X.X.X.X>;tag=A953444-DE8
To: <sip:8007490000@X.X.X.X>
CSeq: 101 INVITE
Content-Length: 0

*Apr 12 14:32:05.255: //528456/956488000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK39D023D9
Record-Route: <sip:X.X.X.X:5060;transport=udp;lr>
Call-ID: 860A521A-E05711E4-A69CF05C-2FA133A4@X.X.X.X
From: "Jaffar AlMoosa"<sip:XXXXXXX@X.X.X.X>;tag=A953444-DE8
To: <sip:8007490000@X.X.X.X>;tag=sbc0804tf2pc227
CSeq: 101 INVITE
Reason: Q.850;cause=65;text="bearer capability not implemented"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0

Thank you,

Shadi

[Eldho E Kuriakose]

But stc is saying pbx problem. I also tested with pri its working fine. You have ttranscoder configured?

[Shadi Shami]

I worked with them a lot and they never said the problem from their side :)

No need to xcoder as we alrerady send the codec they prefer (g711alaw).

Referring to the log file STC provided (traces_ngn-sx-mat-108_sip_2015-04-12-09-36-10.txt) from their proxy server, it's appearing clearly they are disconnecting the call with (488 Not Acceptable Here), so they have to explain why the y disconnecting the call and why other toll free numbers working fine except for this specific one.

Thank you,

Shadi

[Eldho E Kuriakose]

Thanks for your help