Hello,
Failure issue the outbound call.
I want change to the From value.
From: 07044708203<sip:@192.168.117.253>
--> Form: <sip@07044708203@192.168.117.253>
Also the P-Asserted-Identity: <sip:@kt.co.kr>
--> P-Asserted-Identity: <sip:0707700XXXX@kt.co.kr>
Voice class sip-profiles 4 of diversion value is need for a call forwarding.
How do I edit a profile?
321341: *Mar 2 10:13:57: //4207156/C14D12000015/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:01094290707@192.168.117.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.117.xxx:5060;branch=z9hG4bK50C31D10FB
From: 07044708203<sip:@192.168.117.xxx>;tag=76A03F14-11C0
To: <sip:01094290707@192.168.117.xxx>
Date: Mon, 02 Mar 2015 01:13:57 GMT
Call-ID: 3C4518D8-BFB011E4-B8E6DD05-7C94D1C5@192.168.117.253
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3243053568-0000065536-0001402339-3916671168
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1425258837
Contact: <sip:07044708203@192.168.117.xxx:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
P-Asserted-Identity: <sip:@kt.co.kr>
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 305
v=0
o=CiscoSystemsSIP-GW-UserAgent 6598 5639 IN IP4 192.168.117.253
s=SIP Call
c=IN IP4 192.168.117.253
t=0 0
m=audio 17530 RTP/AVP 0 100 101
c=IN IP4 192.168.117.253
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
!
=================================================================================
voice class sip-profiles 4
request INVITE sip-header From modify "\"(.*)\"" ""
request INVITE sip-header P-Asserted-Identity modify "\"(.*)\"" ""
request INVITE sip-header Diversion copy "sip:(.*)@(.*)" u01
request INVITE sip-header From modify "(.*)<sip:(.*)@(.*)>" "\2<sip:\u01@\3>"
request INVITE sip-header P-Asserted-Identity modify "P-Asserted-Identity: (.*)<sip:(.*)@(.*)>" "P-Asserted-Identity: <sip:\u01@kt.co.kr>"
dial-peer voice 2003 voip
description ## Call Forward All Call to KT SBC ##
translation-profile outgoing CFAtoKT
huntstop
destination-pattern ##99.T
rtp payload-type lmr-tone 107
rtp payload-type nte-tone 108
session protocol sipv2
session target ipv4:192.168.117.241
voice-class codec 1
voice-class sip asserted-id pai
voice-class sip profiles 4
dtmf-relay rtp-nte
no vad
!
