Q.850, cause 16 SIP call disconnect problem during call incoming and outgoing

[mohsin majeed]

Dears,

There is disconnect call issue during the call when an CCX agent make or receives a call.

We opened ticket with ITSP, below is the reply from there side.

    Define PCMA as 1^st priority and PCMU as 2^nd priority.

·         Able to send early offer in invite message (We are not receiving early offer inside your invite).

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.200.0.7:5079;branch=z9hG4bKech2udosuasfbfccctshoaeba

Call-ID: t4hppccpkpa2dae7tfub7ho7p4fakse4@SoftX3000

From: <sip:505715017@10.100.200.20;user=phone>;tag=obpu2scc-CC-26

To: <sip:1748240430@10.100.100.20;user=phone>;tag=sbc0803679CAA4C-720

CSeq: 5 INVITE

Date: Sun, 03 May 2015 08:42:15 GMT

Allow: INVITE,OPTIONS,BYE,CANCEL,ACK,PRACK,UPDATE,REFER,SUBSCRIBE,NOTIFY,INFO,REGISTER

Allow-Events: telephone-event

Contact: <sip:1748240430@10.226.190.224:5060>

Supported: replaces,sdp-anat,timer

Server: Cisco-SIPGateway/IOS-15.2.4.M4

Remote-Party-ID: <sip:7731@172.29.46.134>;party=called;screen=no;privacy=off

Content-Length: 229

Content-Type: application/sdp

v=0

o=- 7989 7928 IN IP4 10.201.20.45

s=SBC call

c=IN IP4 10.201.20.45

t=0 0

m=audio 48704 RTP/AVP 0 97

c=IN IP4 10.201.20.45

a=sendonly

a=rtpmap:0 PCMU/8000

a=rtpmap:97 telephone-event/8000

a=fmtp:97 0-15

a=ptime:20

INVITE sip:505715017@10.200.0.7:5079;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.226.190.224:5060;branch=z9hG4bK5EC4654FCT23398

Call-ID: t4hppccpkpa2dae7tfub7ho7p4fakse4@SoftX3000

From: <sip:1748240430@10.100.100.20;user=phone>;tag=sbc0803679CAA4C-720

To: <sip:505715017@10.100.200.20;user=phone>;tag=obpu2scc-CC-26

CSeq: 103 INVITE

Date: Sun, 03 May 2015 08:42:18 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

Max-Forwards: 70

Contact: <sip:1748240430@10.226.190.224:5060>

Expires: 180

Allow-Events: telephone-event

Cisco-Guid: 0158519454-4037480932-2200222861-0419911670

Content-Length: 0

How i can configure these two parameter "early offer" and "PCMA,PCMU"

Your help

[Murali_DS]

u can enforce early offer in Dial peers. Refer the below.

http://www.cisco.com/c/en/us/td/docs/ios/voice/cube/configuration/guide/vb_8241.pdf

Hope this helps.

Thx,

M

[Chris Deren]

Create codec class on the GW with ordered G711alaw and G711ulaw and then apply this class to your dial peers.

[mohsin majeed]

Thanks for your reply,

I also tried as you suggested but no benefit. Now, i configured only one G711alaw.

I attached logs from RTMT may can help to trace the issue.

In debugs the calling number from mobile is 508261660 and called number 8001256666

I will upload the Gateway logs later

[Mukesh Kumar]

Hi mohsin,

Did your issue resolved by configuring early offer?

Or have you done anything else to resolve it?

I am facing the same issue. Call disconnected when the other end pick up the call. In the debugs it shows the same reason cause 16.

Regards,

MUKESH KUMAR | Network Engineer

Spooster IT Services

Computer Networking Solutions

mjindal@spooster.com

www.spooster.com

[Ritesh Desai]

Hey Mukesh Kumar,

FYI... Q.850 Cause Code 16 defines normal clearing. Call disconnection usually refers to Codec mismatch problem. Please cross check what Codec your ISP is sending you G711Alaw or PCMA, G711ULaw or PCMU, G729, etc...

Post this, follow Chris Deren comment.

regards,

Ritesh Desai

[Mukesh Kumar]

Thanks for the information Ritesh.

Codecs are same on both end.

On our end we are using SIP delay offer and on the SIP provider end they are using SIP early offer.

Is this can be the issue of call disconnecting?

Regards,

MUKESH KUMAR | Network Engineer

Spooster IT Services

Computer Networking Solutions

mjindal@spooster.com

www.spooster.com

[Chintan Gajjar]

can you please do ccsip message debug for the problematic call on Voice gateway and post the logs here?

[Ritesh Desai]

At Mukesh Kumar,

Sharing of debug ccsip messages and details of Infra would help to solve the issue.

regards,

Ritesh Desai

[mohsin majeed]

Dear Mukesh,

We opened this case with cisco even. They also replied that this is "normal call clearing". But, believe me this was not normal clearing even there is a message. We (with cisco) couldn't resolve the issue. We tried many things like playing with the codecs and early offer. Our company didn't wait too much because their business based on it. So, finally we had hosted our call center.

But follow the suggestions from community experts; may you have success.

Regards,