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Issues with voice class sip-profiles

Ben Skinner
Level 1
Level 1
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:xxxxxxxx@58.174.xxx.xxx: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:087127xxxx@125.213.160.81:5060>

To: <sip:xxxxxxx@58.174.xxx.xxx:5060>

From: "087127xxxx"<sip:087127xxxx@125.213.160.81:5060>;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:6187200xxxx@125.213.160.81>;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

1 Accepted Solution

Accepted Solutions

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

View solution in original post

3 Replies 3

carunach
Cisco Employee
Cisco Employee

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

Arun

Please see attached file.

Kind Regards,

Ben

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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