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UC540 Incoming Dial Plan

Greetings,

We  have signed up for SIP trunking with Engin.  They provided us with 4  channels and 10 DID's.  The first DID we configured to a Blast group and  this works fine.  The remainder to different extensions, this does not  work.  All numbers dialed ends up ringing the Blast Group?  The Provider  said we must take the called number from the TO field in the SIP  header.  Any ideas?

Many thanks in advance.

Pierre

1 Accepted Solution

Accepted Solutions

Hi,

We use Engin and it works great although you need to do two things. (three speak to someone in Australia otherwise you will go mad!)

1. Tell them you're using a Cisco box so they they add the P-called ID to the SIP header.  (check the SIP debug to ensure its there) - Sometimes if you do things to the account it can drop off

2. add these lines via CLI

no ip nat service sip udp port 5060
no ip nat service sip tcp port 5060

dial-peer voice 1000 voip

voice-class sip call-route p-called-party-id

voice service voip

     sip

      call-route p-called-party-id

You will need to unregiser and re-register the SIP trunk for this to work.

You want the header to look like this in the debug:

INVITE sip:xxx@165.228.188.203:5060 SIP/2.0

Via: SIP/2.0/UDP 203.161.160.71:5060;branch=z9hG4bKktab2j20fo00c99994a1.1

From: "0419316995"<sip:xxxyourregisteredmainnumberxx@voice.mibroadband.com.au;user=phone>;tag=SD4dgjd01-1432174027-1358561378338-

To: "yourname"<sip:xxindialnumberxx@voice.mibroadband.com.au>

Call-ID: SD4dgjd01-5e102a2114280888b03447af02aee062-au418e3

CSeq: 675856402 INVITE

Contact: <sip:SD1o6i6-vv9pmjj9mvp7tbr5iqonkdpvku9rouvrjrdrvorsmqvtoh8gjpv1-6@203.161.160.71:5060;transport=udp>

P-Called-Party-ID: <sip:xxxindialnumberxxx@voice.mibroadband.com.au>

View solution in original post

6 Replies 6

johschaf
Level 4
Level 4

Hello,

The UC will take the inbound number from the TO field in the SIP invite. Can you please post the following:

1. Inbound dial peer

2. Translation profile referenced in the dial peer

3. Translation rules referenced in the translation profile.

Example:

dial-peer voice 3000 voip

description SIP_Inbound

translation-profile incoming SIP_Called

incoming called-number 555100[0-9]

direct-inward-dial

voice translation-profile SIP_Called

translate called 1

voice translation-rule 1

rule 1 /5551000/ /100/

rule 2 /5551001/ /101/

rule 3 /5551002/ /102/

rule 4 /5551003/ /103/

rule 5 /5551004/ /104/

rule 6 /5551005/ /105/

rule 7 /5551006/ /106/

rule 8 /5551007/ /107/

rule 8 /5551008/ /108/

rule 8 /5551009/ /109/

Thanks,

-john

Hi John,

Thanks for the prompt response.  Here are what I have:

dial-peer voice 3000 voip

description Sydney

translation-profile incoming Sydney_Called_4

session protocol sipv2

session target sip-server

incoming called-number 028004130[1-5]

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

!

voice translation-profile Sydney_Called_4

translate calling 3265

translate called 4

!

voice translation-rule 4

rule 1 /0280041301/ /301/

rule 2 /0280041302/ /302/

rule 3 /0280041303/ /303/

rule 4 /0280041304/ /304/

rule 5 /0280041305/ /305/

!

Hope this helps to troubleshoot.

Many thanks

Pierre

Hello,

The dial-peers and translations look correct. Can you setup the following debugs, reproduce the issue, and then post the debug output here.

debug voice ccapi inout

debug ccsip messages

Thanks,

-john

Hi Hohn,

What do you know - it suddenly started working.  Not sure what it was.  Many thanks for your help John!

Kind regards

Pierre

Hi,

We use Engin and it works great although you need to do two things. (three speak to someone in Australia otherwise you will go mad!)

1. Tell them you're using a Cisco box so they they add the P-called ID to the SIP header.  (check the SIP debug to ensure its there) - Sometimes if you do things to the account it can drop off

2. add these lines via CLI

no ip nat service sip udp port 5060
no ip nat service sip tcp port 5060

dial-peer voice 1000 voip

voice-class sip call-route p-called-party-id

voice service voip

     sip

      call-route p-called-party-id

You will need to unregiser and re-register the SIP trunk for this to work.

You want the header to look like this in the debug:

INVITE sip:xxx@165.228.188.203:5060 SIP/2.0

Via: SIP/2.0/UDP 203.161.160.71:5060;branch=z9hG4bKktab2j20fo00c99994a1.1

From: "0419316995"<sip:xxxyourregisteredmainnumberxx@voice.mibroadband.com.au;user=phone>;tag=SD4dgjd01-1432174027-1358561378338-

To: "yourname"<sip:xxindialnumberxx@voice.mibroadband.com.au>

Call-ID: SD4dgjd01-5e102a2114280888b03447af02aee062-au418e3

CSeq: 675856402 INVITE

Contact: <sip:SD1o6i6-vv9pmjj9mvp7tbr5iqonkdpvku9rouvrjrdrvorsmqvtoh8gjpv1-6@203.161.160.71:5060;transport=udp>

P-Called-Party-ID: <sip:xxxindialnumberxxx@voice.mibroadband.com.au>

Hi Tim,

Thanks, this was apparently the solution.  Many thanks for responding.  All is working well after my collegue made these changes.

Kind regards

Pierre