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Issues with SIPnbound dial peer

Dear All,

I am trying to configure my first CME with SIP. 

 

Outbound Calls are working ok, however I am getting fast busy for inbound.

 

I believe my dial peer is okay and I want for testing to translate the number called to the extension 17.

 

Debug gives me this

*Aug 30 12:32:57.146: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number=03303801031, Called Number=03303801031, Peer Info Type=DIALPEER_INFO_SPEECH

*Aug 30 12:32:57.146: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=03303801031

*Aug 30 12:32:57.146: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)

*Aug 30 12:32:57.146: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

   dialstring=03303801031, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1

*Aug 30 12:32:57.146: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

   Result=NO_MATCH(-1)

 

 

My dial peer config is 

voice translation-rule 1

 rule 1 /03303801031/ /17/

!

voice translation-rule 350

 rule 1 /^999$/ /999/

 rule 5 /^9\(.+\)/ /\1/

!

!

voice translation-profile INCOMING

 translate called 1

!

voice translation-profile strip9

 translate called 350

 

 

dial-peer voice 1 voip

 description Outbound 11 digit numbers starting with zerob

 translation-profile outgoing strip9

 destination-pattern ^90[1-9].........

 session protocol sipv2

 session target sip-server

 voice-class codec 1  

 dtmf-relay rtp-nte

 no vad

!

dial-peer voice 2 voip

 description Outbound other numbers

 translation-profile outgoing strip9

 destination-pattern ^9...T

 session protocol sipv2

 session target sip-server

 voice-class codec 1  

 dtmf-relay rtp-nte

 no vad

!

dial-peer voice 999 voip

 description incoming calls 

 translation-profile incoming INCOMING

 session protocol sipv2

 session target sip-server

 incoming called-number 03303801031

 voice-class codec 1  

 dtmf-relay rtp-nte

 no vad

!

1 REPLY

Review this document about

Review this document about dial peers and it should answer any questions you have.

http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/14074-in-dial-peer-match.html

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