Cisco Support Community
cancel
Showing results for 
Search instead for 
Did you mean: 
New Member

Need really urgent help from anyone

Hi ,    

We are implementing sip in our business and we tested the dummy numbers on sip it is working fine.

When we test our main numbers it is not working for incoming or outgoing calls. it is  giving sip error error 404.

I am using CUCM 8.6.2

Any ideas.

Oct 12 03:52:23.198: //418747/863B1AC6BE1B/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x2BCB7F60

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 0861036261

Called Number            : 0893640881

Source IP Address (Sig  ): 123.102.100.34

Destn SIP Req Addr:Port  : 123.102.30.131:5060

Destn SIP Resp Addr:Port : 123.102.30.131:5060

Destination Name         : 123.102.30.131

Oct 12 03:52:23.198: //418747/863B1AC6BE1B/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g711alaw

Negotiated Codec Bytes   : 160

Nego. Codec payload      : 8 (tx), 8 (rx)

Negotiated Dtmf-relay    : 6

Dtmf-relay Payload       : 101 (tx), 101 (rx)

Source IP Address (Media): 123.102.100.34

Source IP Port    (Media): 26776

Destn  IP Address (Media): 58.105.248.129

Destn  IP Port    (Media): 55094

Orig Destn IP Address:Port (Media): [ - ]:0

Oct 12 03:52:23.198: //418747/863B1AC6BE1B/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 1

Disconnect Cause (SIP)   : 404

Oct 12 03:52:27.038: //418717/84AF9822BDCD/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x2BC63A50

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 0478403561

Called Number            : 0893640881

Source IP Address (Sig  ): 123.102.100.34

Destn SIP Req Addr:Port  : 123.102.30.131:5060

Destn SIP Resp Addr:Port : 123.102.30.131:5060

Destination Name         : 123.102.30.131

Oct 12 03:52:27.038: //418717/84AF9822BDCD/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : g711alaw

Negotiated Codec Bytes   : 160

Nego. Codec payload      : 8 (tx), 8 (rx)

Negotiated Dtmf-relay    : 6

Dtmf-relay Payload       : 101 (tx), 101 (rx)

Source IP Address (Media): 123.102.100.34

Source IP Port    (Media): 18150

Destn  IP Address (Media): 58.105.248.129

Destn  IP Port    (Media): 44382

Orig Destn IP Address:Port (Media): [ - ]:0

Oct 12 03:52:27.038: //418717/84AF9822BDCD/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 16

Disconnect Cause (SIP)   : 487

Oct 12 03:52:27.042: //418721/84AF9822BDCD/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x2BCBD950

State of The Call        : STATE_DEAD

TCP Sockets Used         : NO

Calling Number           : 0478403561

Called Number            : 0893640881

Source IP Address (Sig  ): 123.102.100.34

Destn SIP Req Addr:Port  : 123.102.30.131:5060

Destn SIP Resp Addr:Port : 123.102.30.131:5060

Destination Name         : 123.102.30.131

Oct 12 03:52:27.042: //418721/84AF9822BDCD/SIP/Call/sipSPIMediaCallInfo:

Number of Media Streams: 1

Media Stream             : 1

Negotiated Codec         : No Codec

Negotiated Codec Bytes   : 0

Nego. Codec payload      : 255 (tx), 255 (rx)

Negotiated Dtmf-relay    : 0

Dtmf-relay Payload       : 0 (tx), 0 (rx)

Source IP Address (Media): 123.102.100.34

Source IP Port    (Media): 19128

Destn  IP Address (Media):  -

Destn  IP Port    (Media): 0

Orig Destn IP Address:Port (Media): [ - ]:0

Oct 12 03:52:27.042: //418721/84AF9822BDCD/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 16

Disconnect Cause (SIP)   : 487

Oct 12 03:52:27.502: //418722/84F0E7CCBDDA/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x2BC8AFE0

5 REPLIES

Need really urgent help from anyone

HI Rohit,

Please share debug ccsip all for both incoming and outgoing call separately in notepad.

Also please give your call flow i.e from where to where you are calling or called etc..

Regards,

Nishant Savalia

Regards, Nishant Savalia
New Member

Re: Need really urgent help from anyone

Hi Nishant,

Thanks for your reply. I am only testing incoming for the moment. Please see attached notepad for debug information.

Thanks

Regards

Rohit

Re: Need really urgent help from anyone

Hi Rohit,

Logs don't show any kind of messages exchanged. Can you please share "debug ccsip all" or "debug ccsip message".

Also please mention your call flow (e.g. pstn----sip---->gateway-------sip trunk----> cucm). Mention your call flow.

Regards,

Nishant Savalia

Regards, Nishant Savalia
New Member

Re: Need really urgent help from anyone

Hi Nishant,

Call flow

Telco --> Router(CUBE) ---> sip trunk ---> CUCM 8.6.2

Our SIP trunks are configured in cucm. I think call state dead is coming with error 404

Please see attached ccsip all and ccsip message debugs

Thanks

Regards

Rohit

Re: Need really urgent help from anyone

Hi Rohit,

CUCM releasing the call because it's unable to find the called number received i.e. called number does not exist in dial-plan.

Check below points:-

1). Check the CSS on the SIP trunk in CUCM (may be called number is not accessible due to incorrect CSS).

2). Check the translation pattern and it's CSS.

3). If you are not using translation pattern then check for the translation profile in router config. In this case you can check and share your running-config and debug voice dial-peer or debug voice ccapi inout.

Regards,

Nishant Savalia

Regards, Nishant Savalia
509
Views
5
Helpful
5
Replies
CreatePlease to create content