[WXC/VFX] WXC DDI's issues with FreePBX

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Naylz

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Topic # 164302 4-Feb-2015 17:57

Here is my current setup.

I have a SIP Trunk with WXC

I have my pilot number + 10DDI's with WXC

I used to have an openvfx line with WXC but changed to the SIP Trunk because we needed the DDI's to work and to be directed to the correct extension. I have a few entities that work from the same building (all my own entities). The DDI's could not apparently be sent to an openvfx line.

We changed from the openvfx line to the SIP trunk at some expense as well as paid for the transfer of the DDI's per line. With the openvfx line all we could get was the DDI's forwarded to our system and then an IVR that is played to get clients to choose the extension.

I am using an Asterisk based FreePBX system at which I have become extremely proficient.

However the DDI's are not being sent with the DDI's from WXC.

the CID is showing as ‘PILOTNUMBER’ regardless of whether I dial the Main i.e PILOTNUMBER or # or the DDD

This means you are setting the forwarder upstream before it gets to me, which is causing all caller IDs to come through with that number. This is not what DDI’s are. This is just number forwarding.

I presumed that this was they were setting the forwarder upstream before it gets to me, which is causing all caller IDs to come through with that number. This is not what DDI’s are. This is just number forwarding.

they replied

WXC sends the initial Invitation to your Pilot number. We then outline the CID and destination DDI via the 'From' and 'To' fields shown in the test call I have made below.

(replaced actual numbers with DDI and Pilotnumber and changed sip...

INVITE sip:‘PILOTNUMBER’ @IP:5060;transport=udp SIP/2.0
From: <sip:0xxxxxxxxxx@as.wxcnz.net>;tag=96109ab6-13c4-54caefea-958fef2-2d55bcb9
To: "Womens Health Centre Limited" <sip:DDI@as.wxcnz.net;rinstance=5487a13b10796b20>

They suggest that we get FreeBX to look for the DDI information via the 'To' field of our invites.

Is anyone able to do this.

A second issue is that with the openvfx line we could have more than 1 call on the line at any one time and if the lines were busy, the asterix?Freepbpx IVR was activated.

With the SIP Trunk we now realise that we have only 1 channel in ($20) per month and have had to add another channel to this.

If both channels are being used, nothing comes through to the IVR so clients get a busy signal with no answerphone.

We could get voicemail at $5 per DDI or get a 3rd channel at $20 so that if 2 lines are used the 3rd line can go to the Freepbx voicemail.

And we have had to use WXC fax to email service as despite having an ATA we have been unable to get the fax machine to work.

Can anyone offer any solutions to the above problems please.

The most important issue is the DDI"s coming into the office as their proper numbers... Does anyone know how to get the Asterisk/FreePBX system to check the TO field.

or Should I be thinking of a different provider?

Thanks for any advice.

cheers
Naylin

coffeebaron

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Reply # 1231318 4-Feb-2015 18:52

I haven't done this on Asterisk / FreePBX for a while, but do have a couple of Cisco systems setup with this scenario and works no problem. So sorry may not solve you issue, but it will be an issue with configuration at your end.

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sbiddle

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Reply # 1231343 4-Feb-2015 19:54

Did you read my blog post which is the first link on Google?

http://www.geekzone.co.nz/sbiddle/7170

The concurrent call issue isn't an issue is such - it's a bit like only paying for one POTS line, you're only able to make one concurrent call at a time.

maverick

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Reply # 1231465 5-Feb-2015 05:45

The reason the pilot is sent is that this is the number that is registered against the trunk i.e you only need to register the single pilot number to receive all your calls other options have it where you have to register all numbers which is not a good deployment model (a lot of extra sip traffic) , all calls are send to the registered trunk number, the full DDI is in the TO header, this is a standard Invite and normal way of presenting trunk calls, as pointed out this will purely be a configuration issue on your system.

Mr Biddles excellent posts are a pretty good guide to assist as well , as you have only 2 calls for your trunk if its filling up i would suggest that you do need to add the 3rd sim call

Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well

https://www.facebook.com/wxccommunications

breakaway

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Reply # 1232423 6-Feb-2015 22:25

I'm trying to help naylz configure his PBX. I've attempted applying the fix described in sbiddle's post but I'm not getting the desired outcome. I just wanted to check that I'm doing it right:

I have added this code into /etc/asterisk/extensions_custom.conf:

[from-trunk-dvx]
exten => _.,1,Noop(Extract DVX DDI/DID info from SIP URI header. By Steve Biddle stevenbiddle@gmail.com)
exten => _.,n,Noop(NoOp(SIP_HEADER : ${SIP_HEADER})
exten => _.,n,Set(DVXDDI=${SIP_HEADER(To)})
exten => _.,n,Set(DVXDDI=${CUT(DVXDDI,@,1)})
exten => _.,n,Set(DVXDDINAME=${CUT(DVXDDI,",1)})
exten => _.,n,Set(DVXDDI=${CUT(DVXDDI,:,2)})
exten => _.,n,Goto(from-trunk,${DVXDDI},1)

Then I edited my trunk config (Specifically the PEER settings under "Outgoing") to include this line:

context=from-trunk-dvx

But the call is still getting captured by the inbound route for the "Pilot Number". You can view a full sip debug session here -- in this scenario, 091110000 is the number I am calling from for testing, 092220000 is the destination which is coming through correctly in the "TO" field, and 096666666 is the pilot number.

The fix is over nearly 5 years old now, is it possible that the format of how WxC is sending the number has changed & as a result the code is no longer snipping out the DID properly? Unfortunately I don't have enough knowledge with Asterisk to deciper exactly what the snippet is doing. Some asssitance would be much appreciated.

sbiddle

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Reply # 1232496 7-Feb-2015 11:08

You're doing something wrong because inbound calls are still hitting the from-trunk-sip context rather than from-trunk-dvx

breakaway

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Reply # 1232865 8-Feb-2015 12:19

Ok I got it working.

It appears the issue is that in /etc/asterisk/sip_additional.conf, where the SIP trunk details are placed, we had another trunk registered.

It was phsically above the registration info for the DVX trunk (wherein I'd applied the fixes suggested by you steve).

As a result it was never hitting that context -- this is odd, I thought it's supposed to hit the context of the corresponding trunk, not just the first one in the list?

The Solution? I put a "z" in the "Trunk Name" so it's at the bottom of the list now