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New Member

cannot call out from CUCM

Hello, this should be a simple problem to fix as I am only a newbie.

I have a 2801 router running CME which has a SIP connection to my TSP. I have confirmed it is connecting as I have been making calls out and receiving calls. No problems

I have now just added a CUCM 8.6 box which connects to the 2801 router via a SIP trunk.

I am trying (without success) to call out from a phone registered with CUCM via the 2801 gateway.

The calls are leaving CUCM and arriving at the CME gateway but this is where they fail.

I have included the output of a 'debug ccsip call" below, and also attached my CME gateway router config.

I did a "debug voip dialpeer all" and it said the call is matching my outgoing dial peer 1".

Can anyone offer any suggestions as to what is wrong here ? Thanks for any help.

The CUCM phone I am calling from is extension 2001. CUCM is located at 192.168.2.115. I am trying to rin

May 13 07:58:13.763: //57/DB1A1E000000/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x6826C3C0

State of The Call        : STATE_DEAD

TCP Sockets Used         : YES

Calling Number           : 2001

Source IP Address (Sig  ): 192.168.2.1

Destn SIP Req Addr:Port  : 192.168.2.115:5060

Destn SIP Resp Addr:Port : 192.168.2.115:51842

Destination Name         : 192.168.2.115

Router#

May 13 07:58:13.763: //57/DB1A1E000000/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): 192.168.2.1

Source IP Port    (Media): 17714

Destn  IP Address (Media):  -

Destn  IP Port    (Media): 0

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

May 13 07:58:13.763: //57/DB1A1E000000/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 3

Disconnect Cause (SIP)   : 404

  • IP Telephony
4 REPLIES
Bronze

cannot call out from CUCM

These are just hints.  Try to troubleshoot based on the call flow, CUCM-->ROUTER-->TSP, inbound and outbound call legs.  Issue the cmd debug voip ccapi ind 2 and see whether it is hitting the router, and try to issue debug voip dialpeer in out see whether it is going out.  and also see the trunk is registered in cucm. 

Thanks

Hall of Fame Super Silver

cannot call out from CUCM

In your config I dont see IP-IP GW config:

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

What version of IOS are you running, you may be hitting toll-fraud prevention mechanism, use "debug voice ccapi inout" to confirm.

Can you mask your caller ID to valid ANI digits as you are sending 4 digits now to see if that helps? 

Did you try calling the exact same number from CME? 

Can you post "debig voice translation" as your debug shows 9 as part of the called number which would explain the 404 number, perhaps it is not being stripped properly.

HTH

Chris

New Member

Re: cannot call out from CUCM

Hello, thanks.

I turned on "debug voice translation" and "debug ccsip calls". Additonally "debug voice dialpeer" tells me that outgoing dial peer 1 is being matched.

So, "debug ccsip calls" revelas that the leading 9 is not being stripped nor is any negotiated codec being used.

Both of these things should be happening if the outgoing dial peer 1 is being matched.

Also I notice that no incoming dial peer is being matched. Perhaps that is the problem ?

I am unsure how I should correctly configure my incoming dial peer for the call coming from CUCM. Can someone please advise how I should write that ?. The call will be coming across the SIP trunk from a phone with extension 2001.

Thanks for any help.

Here is my output from 'debug dialpeers;:

May 13 20:30:13.137: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number, Called Number=, Peer Info Type=DIALPEER_INFO_SPEECH

May 13 20:30:13.137: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number

May 13 20:30:13.137: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

May 13 20:30:13.137: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

   dialstring=9, saf_enabled=1, saf_dndb_lookup=1, dp_result=0

May 13 20:30:13.137: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0)

   List of Matched Outgoing Dial-peer(s):

     1: Dial-peer Tag=1

May 13 20:30:13.137: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Calling Number=2001, Called Number=, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

May 13 20:30:13.137: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Result=NO_MATCH(-1) After All Match Rules Attempt

May 13 20:30:13.137: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

   dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1

May 13 20:30:13.141: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Calling Number=2001, Called Number=, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

May 13 20:30:13.141: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Result=NO_MATCH(-1) After All Match Rules Attempt

May 13 20:30:13.141: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

   dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1

May 13 20:30:13.141: //-1/E8B7BE000000/DPM/dpAssociateIncomingPeerCore:

Router#Calling Number=2001, Called Number=, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

May 13 20:30:13.141: //-1/E8B7BE000000/DPM/dpAssociateIncomingPeerCore:

   Result=NO_MATCH(-1) After All Match Rules Attempt

May 13 20:30:13.145: //-1/E8B7BE000000/DPM/dpMatchSafModulePlugin:

   dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1

Here are my 'debug voice translation' and 'debug ccsip calls' outputs:

May 13 20:19:44.175: //-1/71CDF5800000/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x6828FEE4; count=1

May 13 20:19:44.175: //-1/71CDF5800000/RXRULE/regxrule_stack_pop_callinfo_internal: numinfo=0x68367EA8

May 13 20:19:44.179: //-1/71CDF5800000/RXRULE/regxrule_get_profile_from_dialpeer_internal: Error: Invalid input peer_tag=0 direction=incoming

May 13 20:19:44.179: //-1/71CDF5800000/RXRULE/regxrule_stack_push_RegXruleNumInfo_internal: stack=0x6828FEE4; count=1

May 13 20:19:44.191: //-1/71CDF5800000/RXRULE/regxrule_stack_pop_RegXruleNumInfo: stack=0x6828FEE4; count=1

May 13 20:19:44.191: //-1/71CDF5800000/RXRULE/regxrule_stack_pop_callinfo_internal: numinfo=0x68367EA8

May 13 20:19:44.191: //224/71CDF5800000/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x67C45D38

State of The Call        : STATE_DEAD

TCP Sockets Used         : YES

Calling Number           : 2001

Called Number            :

Source IP Address (Sig  ): 192.168.2.1

Destn SIP Req Addr:Port  : 192.168.2.115:5060

Destn SIP Resp Addr:Port : 192.168.2.115:33532

Destination Name         : 192.168.2.115

Router#

May 13 20:19:44.191: //224/71CDF5800000/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): 192.168.2.1

Source IP Port    (Media): 18056

Destn  IP Address (Media):  -

Destn  IP Port    (Media): 0

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

May 13 20:19:44.191: //224/71CDF5800000/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC)    : 21

Disconnect Cause (SIP)   : 403

Router#

Re: cannot call out from CUCM

Hi.

Try to add.

dial-peer voice 6 voip

answer-address 2...

voice-class codec 1

dtmf-relay rtp-nte

no vad

HTH

Regards

Carlo

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