Issues with voice class sip-profiles

Answered Question
Sep 20th, 2010
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Hi all, Sorry if this is the wrong place for this.

I'm having an issue with my SIP provider, They seem to add a Diversion on any incoming calls. This makes the incoming call look very annoying on the screen.

"Forward <calling number>
<calling number>
For <called number>
By <called number>"


Received:

INVITE sip:[email protected]:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 125.213.160.81:5060;branch=z9hG4bK17dd5a0514c97f9b6-b0645-0

Max-Forwards: 70

Contact: <sip:[email protected]:5060>

To: <sip:[email protected]:5060>

From: "087127xxxx"<sip:[email protected]:5060>;tag=38527855-co7225-INS001

Call-ID: 14ef-42d-821201001758-img-05-mas-0-125.213.168.6

CSeq: 722501 INVITE

Content-Type: application/sdp

Supported: 100rel

User-Agent: ENSR2.5.4

Content-Length: 451

Diversion: <sip:[email protected]>;reason=unconditional


I have been trying to use a voice class profile to remove this but placing this on my incoming dial peers and my sip does not remove this diversion line.

This is on a UC520 running 150-1.XA3a


voice service voip
sip
  registrar server expires max 3600 min 3600
  localhost dns:icey.mine.nu
  no update-callerid
  sip-profiles 1


!
voice class sip-profiles 1
request INVITE sip-header Diversion remove
request ANY sip-header Diversion remove


Debug of voice dialpeer inout:


014012: Sep 21 00:17:22.344: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=2633



Dial Peers:


dial-peer voice 2633 voip
corlist outgoing call-domestic
description ** Australian Domestic Pattern via SIP **
translation-profile outgoing SIP_Outgoing
destination-pattern 0[2-9].......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip profiles 1
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 2901 voip
description ** Inbound Dial Peer - SIP **
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
session protocol sipv2
session target sip-server
incoming called-number .%
voice-class codec 1
voice-class sip profiles 1
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad


Any one have any ideas?


Regards,

Ben

Correct Answer by carunach about 6 years 8 months ago

The SIP profiles feature applies for outgoing SIP messages. In your scenario, the inbound SIP INVITE is not routed to another SIP endpoint across a voip dial-peer (i.e. it is not a SIP->SIP CUBE  or TDM->SIP call flow) and hence the SIP profile is not taking effect.


We can use translation profile to remove the redirect number :


voice translation-rule 1

  rule 1 /61872001234/ //


voice translation-profile strip-redirect

   translate redirect-called 1


dial-peer voice 2901

  translation-profile incoming strip-redirect


Arun

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carunach Wed, 09/22/2010 - 11:49
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  • Cisco Employee,

Configuration looks fine. Can you please collect "debug cssip all" along "debug voip ccapi inout" during low call volume?


Arun

Correct Answer
carunach Fri, 09/24/2010 - 07:42
User Badges:
  • Cisco Employee,

The SIP profiles feature applies for outgoing SIP messages. In your scenario, the inbound SIP INVITE is not routed to another SIP endpoint across a voip dial-peer (i.e. it is not a SIP->SIP CUBE  or TDM->SIP call flow) and hence the SIP profile is not taking effect.


We can use translation profile to remove the redirect number :


voice translation-rule 1

  rule 1 /61872001234/ //


voice translation-profile strip-redirect

   translate redirect-called 1


dial-peer voice 2901

  translation-profile incoming strip-redirect


Arun

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