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How to get SIP trunk to use To header instead of Invite header

Ruud van Strijp
Level 1
Level 1

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

3 Replies 3

dijohn
Cisco Employee
Cisco Employee

Can you send in the sh run.

Where are these translation profiles used?

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

!

I found the way to do it. I need to use a 'voice class uri sip'. This is the configuration I used for my situation:

!

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

!