12-30-2010 02:24 AM - edited 03-16-2019 02:38 AM
Hello,
We have a SIP trunk to our UC540 that works fine for it's main number (xxxxx689). However, when I call the secondary number (xxxxx973) on the line, the conversation gets delivered to the phone that is registered with the first number on the line. To me, the problem seems to be in the fact that the called number (xxxxx973) is only to be found in the To header of the SIP message: In the Invite header the main number (xxxxx689) is sent.
[quote]INVITE sip:xxxxx689@172.*.*.25:5060 SIP/2.0
Record-Route: <sip:213.*.*.68;lr=on;ftag=95875AC-223B>
Via: SIP/2.0/UDP 213.*.*.68;branch=z9hG4bKd0d2.ea666783.0
Via: SIP/2.0/UDP 213.*.*.69;branch=z9hG4bKd0d2.3fa15795.0
Via: SIP/2.0/UDP 212.*.*.245:5060;x-route-tag="cid:DMS-TRUNK@212.*.*.245";branch=z9hG4bK2ED4924F5
From: <sip:xxxxx144@test.*.com>;tag=95875AC-223B
To: <sip:xxxxx973@213.*.*.69>
Call-ID: 78794E10-E8A11E0-B3518A20-8F591572@212.*.*.245
Supported: 100rel,timer,replaces
Min-SE: 1800
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 68
Contact: <sip:xxxxx144@212.*.*.245:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 306
To-Hint: sip:xxxxx689@test.*.com
From-Hint: sip:xxxxx144@test.*.com
From-Number-Hint: sip:xxxxx144@test.*.com
P-hint: usrloc applied
Remote-Party-ID: <sip:xxxxx144@test.*.com>;party=calling;screen=yes[/quote]
When I look at the SIP trace of a different (completely working) trunk, I see that the subnumbers on the trunk get sent with the right number in the Invite and the To header. Based on this, I assume our UC540 uses the Invite header instead of the To header to judge where to send the conversation.
Also, the right translation-rule exists:
[quote]!
voice translation-rule 12
rule 1 /xxxxx689/ /824/
rule 2 /xxxxx973/ /800/
!
[/quote]
When I change the 824 to another number, it gets diverted to there. However, when I call the xxxxx973 number, I still reach the 824 number instead of 800 (voicemail).
So, is there a way to let my UC540 use the number in the To header instead of the Invite header? Or is this really something my SIP provider does wrong?
Thanks,
Ruud van Strijp
12-30-2010 02:35 AM
Can you send in the sh run.
Where are these translation profiles used?
12-30-2010 03:08 AM
Thanks for your reply. The config is quite a mess, since it's made with CCA (like Cisco advises to do with the UC series) and then manually tweaked.
As a little summary, here are the rules:
!
voice translation-rule 11
rule 1 /^/ /000/
!
!
voice translation-rule 12
rule 1 /xxxxx689/ /824/
rule 2 /xxxxx973/ /800/
!
!
voice translation-profile PSTN-IN
translate calling 11
translate called 12
!
!
dial-peer voice 1000 voip
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
translation-profile incoming PSTN-IN
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
01-03-2011 07:28 AM
I found the way to do it. I need to use a 'voice class uri
!
voice class uri SIP689 sip
user-id 689
!
voice class uri SIP973 sip
user-id 973
!
voice translation-rule 689
rule 1 /.*/ /823/
!
voice translation-rule 973
rule 1 /.*/ /824/
!
!
voice translation-profile PSTN-IN689
translate calling 11
translate called 689
!
!
voice translation-profile PSTN-IN973
translate calling 11
translate called 973
!
!
dial-peer voice 1000 voip
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
translation-profile incoming PSTN-IN689
session protocol sipv2
session target sip-server
incoming called-number 31xxxxxx689
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
incoming uri to SIP689
no vad
!
dial-peer voice 1001 voip
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
translation-profile incoming PSTN-IN973
session protocol sipv2
session target sip-server
incoming called-number 31xxxxxx973
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
incoming uri to SIP973
no vad
!
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