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- Jan 2, 2008
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Hi, I am having a lot of problems with REVIEW INVITE source errors for my DDI numbers. I'll details as much as I can but if there is anything obvious anyone thinks I have missed please let me know and I'll post the extra. I'm not sure if I have found a bug or just missing something obvious.
We have one VOIP provider (VOIP Unlimited) providing our main telephone number 01695555285 with a block of associated DDI numbers (lets say 01695111500 to 01695111510). We have setup this provider and they work fine, we have also setup all the DDI numbers with associated inbound rules.
Our 3CX system has always been on a server wiith a public IP address (let's call this 111.111.111.111 for now) so the only way we were able to get our DDI ranges to trigger a source identification rule was to add an "Inbound Parameters > Call Source Identification" rule against the VOIP Provider. This rules reads ...
Match Any Fields
Request Line URI: Host Part, Custom Field, 111.111.111.111:5060
and this works fine because all our SIP traffic comes in on this IP and the host part of the INVITE request is always 111.111.111.111:5060. However, now we are going to use a new router which is going to have two public IP connections (let's says 222.222.222.222 and 333.333.333.333) and and is going to NAT the outbound traffic from the 3CX server which now has a local IP (lets say it's 192.168.1.1). The problem is that the incoming SIP requests may have two different public IP address and the "Request Line URI: Host Part, Custom Field" will not accept two parameters and I don't know which IP the message is going to come in on as the router is controlling which link is up depending on failover etc.
I then tried abandoning the idea of using an inbound cal source identification and decided to try using the DDI option in the VOIP Provider. I listed all my DID numbers in the relevant boxes and then selected the "SIP Field containing DID numbers" (which was currently blank) and chose "Request Line URI: User Part" which is where the incoming DID number is placed by my VOIP provider but this has not worked. Also because this field was previously blank and is now set to "User Part" I now cannot remove it and adding back in my inbound parameter source identification also has no effect so i think there may be a problem here.
Here is a SIP message from a failed lookup ...
14:13:59.136 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected]:45169;rinstance=d3e7c28b7f6b8eee SIP/2.0
Via: SIP/2.0/UDP 91.151.2.130;branch=z9hG4bK47f6.b50cb0e5.0
Via: SIP/2.0/UDP 91.151.11.20:5060;rport=5060;branch=z9hG4bK2e95447d79d598a25cebf673da9837f6
Max-Forwards: 67
Record-Route: <sip:91.151.2.130;lr=on;ftag=3441625669-741784>
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=3441625669-741784
Call-ID: 1086178-3441625669-741778@msx2-voip-unl ... domain.com
CSeq: 1 INVITE
Session-Expires: 3600;refresher=uac
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Call-Info: <sip:91.151.11.20>;method="NOTIFY;Event=telephone-event;Duration=1000"
Supported: timer, 100rel
Allow-Events: telephone-event
Content-Length: 0
Any help would be very greatly apprecaited. Many thanks, Dominic Batty
We have one VOIP provider (VOIP Unlimited) providing our main telephone number 01695555285 with a block of associated DDI numbers (lets say 01695111500 to 01695111510). We have setup this provider and they work fine, we have also setup all the DDI numbers with associated inbound rules.
Our 3CX system has always been on a server wiith a public IP address (let's call this 111.111.111.111 for now) so the only way we were able to get our DDI ranges to trigger a source identification rule was to add an "Inbound Parameters > Call Source Identification" rule against the VOIP Provider. This rules reads ...
Match Any Fields
Request Line URI: Host Part, Custom Field, 111.111.111.111:5060
and this works fine because all our SIP traffic comes in on this IP and the host part of the INVITE request is always 111.111.111.111:5060. However, now we are going to use a new router which is going to have two public IP connections (let's says 222.222.222.222 and 333.333.333.333) and and is going to NAT the outbound traffic from the 3CX server which now has a local IP (lets say it's 192.168.1.1). The problem is that the incoming SIP requests may have two different public IP address and the "Request Line URI: Host Part, Custom Field" will not accept two parameters and I don't know which IP the message is going to come in on as the router is controlling which link is up depending on failover etc.
I then tried abandoning the idea of using an inbound cal source identification and decided to try using the DDI option in the VOIP Provider. I listed all my DID numbers in the relevant boxes and then selected the "SIP Field containing DID numbers" (which was currently blank) and chose "Request Line URI: User Part" which is where the incoming DID number is placed by my VOIP provider but this has not worked. Also because this field was previously blank and is now set to "User Part" I now cannot remove it and adding back in my inbound parameter source identification also has no effect so i think there may be a problem here.
Here is a SIP message from a failed lookup ...
14:13:59.136 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected]:45169;rinstance=d3e7c28b7f6b8eee SIP/2.0
Via: SIP/2.0/UDP 91.151.2.130;branch=z9hG4bK47f6.b50cb0e5.0
Via: SIP/2.0/UDP 91.151.11.20:5060;rport=5060;branch=z9hG4bK2e95447d79d598a25cebf673da9837f6
Max-Forwards: 67
Record-Route: <sip:91.151.2.130;lr=on;ftag=3441625669-741784>
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=3441625669-741784
Call-ID: 1086178-3441625669-741778@msx2-voip-unl ... domain.com
CSeq: 1 INVITE
Session-Expires: 3600;refresher=uac
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Call-Info: <sip:91.151.11.20>;method="NOTIFY;Event=telephone-event;Duration=1000"
Supported: timer, 100rel
Allow-Events: telephone-event
Content-Length: 0
Any help would be very greatly apprecaited. Many thanks, Dominic Batty