01-19-2016 06:21 AM - edited 03-17-2019 05:33 AM
Hi all,
I have a problem routing inbound Sip calls to correct extensions in this scenario:
- Cisco 2921 CME IOS vers.15.4(2)T, CME version 10.0
- Sip User agent registered to provider:
sip-ua
credentials username 5406031279 password 7 xxxxxxxxxxxxxxxxx realm messagenet.it
authentication username 5406031279 password 7 xxxxxxxxxxxxxxxxxx
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 5
timers register 200
registrar dns:sip.messagenet.it:5060 expires 300
sip-server dns:sip.messagenet.it
notify telephone-event max-duration 3000
connection-reuse
host-registrar
!
- Bound to the above Uri (5406031279) are several public numbers (i.e. 0292969101 etc.)
translation rules and incoming dial peer, are currently set as follow, because provider pass me the URI on the field "To"
and pass the public number in the userpart of the header "To"
!
voice translation-rule 4
rule 1 /^\(.*\)/ /0\1/
!
!
voice translation-rule 7
rule 1 /5406031279/ /399/
!
!
voice translation-profile RENUM-IN-PROVV
translate calling 4
translate called 7
!
dial-peer voice 1000 voip
description TRUNK SIP KPN
translation-profile incoming RENUM-IN-PROVV
translation-profile outgoing profile1
destination-pattern 0T
redirect ip2ip
session protocol sipv2
session target dns:sip.messagenet.it
incoming called-number .%
voice-class codec 2
dtmf-relay rtp-nte
no vad
!
In this case, calls to all numbers, all go to dial peer 399 that lead to a voice hunt group.
what I want to do is to be able to read the content of userpart of header "To" and
implement a rule or something that allow me to reach the extension directly.
an ephone-dn as example is:
!
ephone-dn 11 dual-line
number 0292969100 secondary 200 no-reg both
pickup-group 1
label 200.Monica
description 200.Monica
name +390292969100
call-forward noan 401 timeout 30
hold-alert 30 originator
!
thanks a lot in advance
Emanuele
01-19-2016 07:25 AM
Hi,
Can you post a "debug ccsip messages" of the SIP call so that we can see exactly what the provider is sending you.
01-19-2016 07:57 AM
Hi James,
this is debug ccsip messages you asked me for, calling number is +393484160884 and I've called +390292969196.
thanks a lot
Jan 19 15:34:45.121: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
...
Jan 19 15:34:47.145: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
Jan 19 15:34:56.457: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:5406031279@100.68.23.82:5060 SIP/2.0
Record-Route: <sip:212.97.59.76;lr=on;ftag=as71dbda44;rpp=np>
Via: SIP/2.0/UDP 212.97.59.76;branch=z9hG4bKcabe.857f13e7.0
Via: SIP/2.0/UDP 193.227.104.46:5060;branch=z9hG4bK07e4e24b
Max-Forwards: 69
From: "+393484160884" <sip:+393484160884@sip.messagenet.it>;tag=as71dbda44
To: <sip:0292969196@212.97.59.76>
Contact: <sip:+393484160884@193.227.104.46:5060>
Call-ID: 54f9b13d2cb5eba93828833016685e66@sip.messagenet.it
CSeq: 102 INVITE
User-Agent: oscar
Date: Tue, 19 Jan 2016 15:34:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 369
v=0
o=root 1309115217 1309115217 IN IP4 193.227.104.39
s=Messagenet
c=IN IP4 193.227.104.39
t=0 0
m=audio 36582 RTP/AVP 18 3 97 8 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
Jan 19 15:34:56.481: //714648/06E0E1519597/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.97.59.76;branch=z9hG4bKcabe.857f13e7.0,SIP/2.0/UDP 193.227.104.46:5060;branch=z9hG4bK07e4e24b
From: "+393484160884" <sip:+393484160884@sip.messagenet.it>;tag=as71dbda44
To: <sip:0292969196@212.97.59.76>
Date: Tue, 19 Jan 2016 15:34:56 GMT
Call-ID: 54f9b13d2cb5eba93828833016685e66@sip.messagenet.it
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.2.T
Content-Length: 0
Jan 19 15:34:56.549: //714648/06E0E1519597/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.97.59.76;branch=z9hG4bKcabe.857f13e7.0,SIP/2.0/UDP 193.227.104.46:5060;branch=z9hG4bK07e4e24b
From: "+393484160884" <sip:+393484160884@sip.messagenet.it>;tag=as71dbda44
To: <sip:0292969196@212.97.59.76>;tag=9CE8E738-2494
Date: Tue, 19 Jan 2016 15:34:56 GMT
Call-ID: 54f9b13d2cb5eba93828833016685e66@sip.messagenet.it
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:5406031279@100.68.23.82:5060>
Record-Route: <sip:212.97.59.76;lr=on;ftag=as71dbda44;rpp=np>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.4.2.T
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 4267 6147 IN IP4 100.68.23.82
s=SIP Call
c=IN IP4 100.68.23.82
t=0 0
m=audio 23386 RTP/AVP 0 101
c=IN IP4 100.68.23.82
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Jan 19 15:34:56.557: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:5406031279@100.68.23.82:5060 SIP/2.0
Record-Route: <sip:212.97.59.76;lr=on;ftag=as71dbda44;rpp=np>
Via: SIP/2.0/UDP 212.97.59.76;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 193.227.104.46:5060;branch=z9hG4bK054b9b61
Max-Forwards: 69
From: "+393484160884" <sip:+393484160884@sip.messagenet.it>;tag=as71dbda44
To: <sip:0292969196@212.97.59.76>;tag=9CE8E738-2494
Contact: <sip:+393484160884@193.227.104.46:5060>
Call-ID: 54f9b13d2cb5eba93828833016685e66@sip.messagenet.it
CSeq: 102 ACK
User-Agent: oscar
Content-Length: 0
Jan 19 15:35:02.229: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
REGISTER sip:192.168.100.1 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.137:5060;branch=z9hG4bKa15f3c6aa8a38ecbf6444aaa502446aa;rport
From: "0292969156" <sip:0292969156@192.168.100.1>;tag=2904050772
To: "0292969156" <sip:0292969156@192.168.100.1>
Call-ID: 389922895@10_10_10_137
CSeq: 4501 REGISTER
Contact: <sip:0292969156@10.10.10.137:5060>
Max-Forwards: 70
User-Agent: A510 IP/42.076.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
Jan 19 15:35:02.233: //714660/0A51A07095B3/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.137:5060;branch=z9hG4bKa15f3c6aa8a38ecbf6444aaa502446aa;rport
From: "0292969156" <sip:0292969156@192.168.100.1>;tag=2904050772
To: "0292969156" <sip:0292969156@192.168.100.1>;tag=9CE8FD6C-664
Date: Tue, 19 Jan 2016 15:35:02 GMT
Call-ID: 389922895@10_10_10_137
Server: Cisco-SIPGateway/IOS-15.4.2.T
CSeq: 4501 REGISTER
Contact: <sip:0292969156@10.10.10.137:5060>;expires=180
Expires: 180
Content-Length: 0
Jan 19 15:35:08.641: //714648/06E0E1519597/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:+393484160884@193.227.104.46:5060 SIP/2.0
Via: SIP/2.0/UDP 100.68.23.82:5060;branch=z9hG4bK1326415C4
From: <sip:0292969196@212.97.59.76>;tag=9CE8E738-2494
To: "+393484160884" <sip:+393484160884@sip.messagenet.it>;tag=as71dbda44
Date: Tue, 19 Jan 2016 15:34:56 GMT
Call-ID: 54f9b13d2cb5eba93828833016685e66@sip.messagenet.it
User-Agent: Cisco-SIPGateway/IOS-15.4.2.T
Max-Forwards: 70
Route: <sip:212.97.59.76;lr=on;ftag=as71dbda44;rpp=np>
Timestamp: 1453217708
CSeq: 101 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=597,OS=95520,PR=604,OR=96640,PL=0,JI=0,LA=0,DU=12
Content-Length: 0
Jan 19 15:35:08.653: //714648/06E0E1519597/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 100.68.23.82:5060;rport=5060;received=151.0.246.147;branch=z9hG4bK1326415C4
Record-Route: <sip:212.97.59.76;lr=on;ftag=9CE8E738-2494;rpp=pn>
From: <sip:0292969196@212.97.59.76>;tag=9CE8E738-2494
To: "+393484160884" <sip:+393484160884@sip.messagenet.it>;tag=as71dbda44
Call-ID: 54f9b13d2cb5eba93828833016685e66@sip.messagenet.it
CSeq: 101 BYE
Server: oscar
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
Jan 19 15:35:10.177: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
...
01-19-2016 08:17 AM
Hi,
I have never had to do this and I would first ask the SIP provider whether they can modify the INVITE messages so that they use the number that you want.
If they cannot do this then you might be able to use SIP Normalization with SIP profiles to modify the messages.
I have never used them to do something as complex as this though so I am not sure whether it would work.
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