cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
36486
Views
0
Helpful
17
Replies

SIP/2.0 404 Not Found Error on Incoming Calls

Abayomi Smith
Beginner
Beginner

Hi All,

I would appreciate any help concerning this issue.

I am unable to make or receive calls. I receive SIP/2.0 404 Not Found on incoming call.

My setup is as thus;

CUCM-----------CUBE----------SIP Gateway

 

Please find attached part of the config of the CUBE and also output of debug ccsip message of test call from 02071871717 to 02038897551.

 

Regards,

Y.

 

 

1 Accepted Solution

Accepted Solutions

Called Number=02038897551. The significant digits field is set to ALL.​

So, you have translation pattern configured to convert the called number to internal extension? 

 

How are the internal extensions configured and assigned to the phones? 

//Suresh Please rate all the useful posts.

View solution in original post

17 Replies 17

Suresh Subramanian
Rising star
Rising star

Hi, it seems like issue with your dial-peer config.

 

can you try configure the below dial-peers and let me know how it works?

 

also you should try binding the right interfaces in the dial-peers

 

voice-class sip bind control source-interface <LAN interface towards CUCM ex: gig0/0>
 voice-class sip bind media source-interface <LAN interface towards CUCM ex: gig0/0>

 

!
dial-peer voice 1000 voip
 description **Inbound call from SIP Trunk**
 session protocol sipv2
 incoming called-number .
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!

!
dial-peer voice 1000 voip
 description **Inbound call from SIP Trunk**
 session protocol sipv2
 incoming called-number .
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!

!
dial-peer voice 1001 voip
 description **Inbound call to CUCM**
 destination-pattern 020388975[0-6].
 session protocol sipv2
 session target ipv4:10.127.33.3
 voice-class sip bind control source-interface <LAN interface towards CUCM ex: gig0/0>
 voice-class sip bind media source-interface <LAN interface towards CUCM ex: gig0/0>
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!


dial-peer voice 1002 voip
 description **Outgoing Call to SIP Trunk**
 destination-pattern .T
 session protocol sipv2
 session target ipv4:88.XX.XX.169
voice-class sip bind control source-interface <WAN interface towards ITSP ex: gig0/1>
 voice-class sip bind media source-interface <WAN interface towards ITSP ex: gig0/1>
 dtmf-relay rtp-nte
 codec g711alaw
 no vad

 

//Suresh Please rate all the useful posts.

Thanks Suresh,

However I am still unable to get any calls through. Instead now I get nothing...Just silence. No ringback nothing.

When I check the debug ccsip messages I get a SIP/2.0 503 Service Unavailable-No Bandwidth Available and also a SIP/2.0 500 Internal Server Error.

 

When I did a debug of the dial-peer just to check if the right dial peer is being matched, I get this

 

dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
*Nov 20 19:36:06.650 GMT: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=02038897500T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Nov 20 19:36:06.650 GMT: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/DPM/dpMatchCore:
   Result=-1
*Nov 20 19:36:06.650 GMT: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ORIGINATE; Calling Number=02038897500
*Nov 20 19:36:06.650 GMT: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
*Nov 20 19:36:06.650 GMT: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=02038897500T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Nov 20 19:36:06.650 GMT: //-1/xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx/DPM/MatchNextPeer:
   Result=Success(0); Incoming Dial-peer=1002 Is Matched
*Nov 20 19:36:06.650 GMT: //-1/4DCAD934-7023-11E4-93CF-FBC03BACF4BB/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_VIA_URI; URI=sip:88.XX.XX.169:5060
*Nov 20 19:36:06.650 GMT: //-1/4DCAD934-7023-11E4-93CF-FBC03BACF4BB/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
*Nov 20 19:36:06.650 GMT: //-1/4DCAD934-7023-11E4-93CF-FBC03BACF4BB/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Nov 20 19:36:06.650 GMT: //-1/4DCAD934-7023-11E4-93CF-FBC03BACF4BB/DPM/dpMatchCore:
   Result=-1

This shows its matching dial-peer 1002, which is the outgoing dial-peer, instead of 1000.

 

Also  the calling number seems to be the first number in the DDI range 02038897500 however I am calling from 02071871717.

I don't have redirect header on inbound calls ticked on the sip trunk config. I believe it should show the originating no is the URI

 

Thanks,

Y

 

Hi.

Just few questions

Did you ckeck that you correctly configured trunk destination address on CUCM?

Can incoming CSS configured on trunk reach internal extensions?

 

Can you please add these lines on your cube:

voice service voip

no ip address trusted authenticate 

 

 

Please add also the output of a debug ccdip mess while calling outbound

 

Thanks 

 

 

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"

Thanks Carlo,

The CSS configured for inbound calls on the trunk can reach the partition of internal extensions.

 

I added the command no ip address trusted authenticate   but there was still no luck. will post the debug ccsip messages output on external calls soon.

 

Thanks,

Y