01-22-2013 02:24 PM - edited 03-21-2019 06:52 AM
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
Solved! Go to Solution.
01-25-2013 04:32 PM
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>
01-22-2013 03:42 PM
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
01-22-2013 07:48 PM
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
01-23-2013 09:58 AM
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
01-23-2013 09:43 PM
Hi Hohn,
What do you know - it suddenly started working. Not sure what it was. Many thanks for your help John!
Kind regards
Pierre
01-25-2013 04:32 PM
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>
01-28-2013 06:03 PM
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
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