ā09-20-2010 06:23 PM - edited ā03-17-2019 10:06 PM
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
Solved! Go to Solution.
ā09-24-2010 07:42 AM
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
ā09-22-2010 11:49 AM
Configuration looks fine. Can you please collect "debug cssip all" along "debug voip ccapi inout" during low call volume?
Arun
ā09-24-2010 01:15 AM
ā09-24-2010 07:42 AM
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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