SIP/2.0 403 Forbidden - Outgoing SIP trunk call getting failed

[mohsin majeed]

Hi Experts,

We have SIP trunk and E1 in our environment. CSS for SIP and E1 are created seperately.

On the SIP trunk we are facing outgoing call failed issue. Route Pattern for the SIP-Local for outgoing call is created to call within the city. Call is successful, but some local number in the same city are not dial able; gives busy signal. These same non-working number can be dialed from E1. This problem is only on the SIP trunk not E1.

Calling number : 1515

Called number : 7 digits Local numbers, 01 is access code for local dialing. 014193193(working) ; 014565235 and 014791233 (non-working)

Traces are attached with file name

Like above the national call to other cities are successful but for some other cities is not working. 

Calling number : 1515

Called number : 10 digits national numbers, 02 is access code for national dialing.

(non-working cities) 020126480008 and 020138267888 ; (working cities) 020163250488 and  020125374003

CM Route pattern --> SIP_Outgoing-RL (Three sip trunks toward VGW with 172.18.0.10 ; 10.14.22.1 ; 10.14.22.5 destination IPs respectively.)

Route Pattern, VGW configu, Numbers and Traces are attached for working and non working calls to Local and National number. In the traces i can see there is SIP/2.0 403 Forbidden 

Appreciate your help

Thanks

[MOHIT SINGH]

Hi Mohsin,

As per the logs, the issue seems to be from provider side. They are sending below reason to disconnect the call for call to "4565235" but same capabilities (G711 ulaw and SIP udp) work for "4193193"

"Reason: Q.850;cause=57;text="bearer capability not authorized"

What is the difference in above 2 numbers? kindly check with provider why the are responding with above cause code.

[mohsin majeed]

Hi Mohit,

Thanks for your time.

It seems to me also, the issue from provider where one number is being dialed and second is not. I will check with provider and update. 

Regards,

[mohsin majeed]

Hi Mohit,

Here are findings from ITSP. As per them we have to send two codecs PCMA and PCMU as 1st and 2nd priority. Under the outgoing dial peer configuration i have codec class with both. Where i can check in the CM to send both the codecs. Gateway config is attached

From ITSP 

We checked and found that there is  the issue with Voice Codec negotiation. Your SIP Voice Gateway is sending only PCMA.

Please make the sure that to make PCMA as 1^st Priority and PCMU as 2^nd Priority.

Your SIP Call from “4565235”

INVITE sip:4565235@10.200.0.7:5060 SIP/2.0 Via: SIP/2.0/UDP 10.226.190.251:5060;branch=z9hG4bK7DDB1351T26246 Record-Route: <sip:10.226.190.251:5060;transport=udp;lr> Call-ID: isbc4BACC511-62011E6-970BAEFE-12F59962@172.29.49.82 From: "Mohsin Majeed"<sip:2905400@10.200.0.7>;tag=sbc08042D8C06D4-1CD1 To: <sip:4565235@10.200.0.7> CSeq: 101 INVITE Date: Wed, 20 Apr 2016 11:17:25 GMT Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 User-Agent: Cisco-SIPGateway/IOS-15.2.4.M4 Allow: INVITE,OPTIONS,BYE,CANCEL,ACK,PRACK,UPDATE,REFER,SUBSCRIBE,NOTIFY,INFO,REGISTER Contact: <sip:2905400@10.226.190.251:5060> Expires: 180 Allow-Events: telephone-event Max-Forwards: 68 Session-Expires: 86400 Remote-Party-ID: "Mohsin Majeed" <sip:2905400@172.29.49.82>;party=calling;screen=yes;privacy=off Cisco-Guid: 3041028736-0000065536-0000030215-0202968842 Content-Length: 220 Content-Type: application/sdp Content-Disposition: session;handling=required v=0 o=- 6371 4018 IN IP4 10.201.20.45 s=SBC call c=IN IP4 10.201.20.45 t=0 0 m=audio 44320 RTP/AVP 0 101 c=IN IP4 10.201.20.45 a=rtpmap:0 PCMU/8000                   ==== 2nd Priority (Make PCMA 1st Priority) a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20

SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.226.190.251:5060;branch=z9hG4bK7DDB1351T26246 Call-ID: isbc4BACC511-62011E6-970BAEFE-12F59962@172.29.49.82 From: "Mohsin Majeed"<sip:2905400@10.200.0.7>;tag=sbc08042D8C06D4-1CD1 To: <sip:4565235@10.200.0.7> CSeq: 101 INVITE Content-Length: 0

SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.226.190.251:5060;branch=z9hG4bK7DDB1351T26246 Call-ID: isbc4BACC511-62011E6-970BAEFE-12F59962@172.29.49.82 From: "Mohsin Majeed"<sip:2905400@10.200.0.7>;tag=sbc08042D8C06D4-1CD1 To: <sip:4565235@10.200.0.7>;tag=ubch7aho CSeq: 101 INVITE Reason: Q.850;cause=57;text="bearer capability not authorized" Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR" Content-Length: 0

Please make sure that these VOICE codecs are configured as per the stated above

[MOHIT SINGH · Accepted solution]

Hi Mohsin,

I will recommend to test 2 things.

1. In CUCM SIP trunk, enable MTP required and set codec to G711alaw. Make an outbound call and test if it works? Also confirm in debug the outgoing codec is G711 alaw in SIP Invite to ITSP.

2. If above step does not work then in CUCM SIP trunk, disable "MTP required" if it is enabled. Also disable "Early Offer support for voice and video calls (insert MTP if needed)" if it's enabled in SIP profile assigned to SIP trunk. Make calls and test. Check what codec is forwarded to ITSP.

[Deepak Mehta]

You can change codec priority order in gateway to match what service provider is asking for under voice-class codec 

On cucm region settings between phone and gateway sip trunk will decide what codec is being negotiated.

[mohsin majeed]

Hi deepak, 

ITSP asked 711alaw as 1st priority. This codec class configuration is already there in gateway. Issue was on the SIP trunk codec priority as attached in comment.

Thanks for the suggestion

[mohsin majeed]

Hi Mohit,

I managed to resolve this issue by just making 711alaw as 1st priority under sip trunk, attached.

[Deepak Mehta]

Thank you for correcting me i might have overlooked the logs.