No outbound calls. UK AQL Trunk.

[jddixon]

Hi all

I am having major problems getting my first install of FusionPBX to call outside of the PBX via the SIP trunk to the upstream provider (AQL).

I am very new to running a VoIP server of any kind but I have a little hands-on experience with 3CX and old versions of Elastix (prior to the move to 3CX). I have set up 1 domain on Fusion, 1 extension and have inbound calls working fine, I can even call back into the PBX from the extension that has been set up and call other ring groups etc. I however have tried almost everything to get the outbound SIP trunk working over the last 2 weeks and still coming up blank. I thought it may be related to the regular expression but I believe this should now be correct because instead of instantly failing, it now just hangs for 20ish seconds until it times out.

If anyone can help it would be greatly appreciated. If not I will need to find a consultant to unpick the issue. I have reached out to the trunk provider (AQL) who were zero help!

I have uploaded the output of siptrace to pastebin to save space here.

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pastebin.com

Thanks in advance

Jon

 

[DigitalDaz]

Your outbound route is bad, you are only sending a 0:

INVITE sip:0@141.170.24.20:5080 SIP/2.0

If they want 44 format, try ^0(.*)$ with a prefix of 44

 

[jddixon]

Hi @DigitalDaz

Thanks so much for having a look. It appears I am now sending out the right details (posted below) but I am now getting another error (501, not implemented).

INVITE sip:447588743157@siptrunking.sbc.aql.com SIP/2.0

I also noticed this in the log file at the same time the connection below. Only problem is its not related to any extensions or IP addresses we have here or at the provider. Could this be a problem or is it someone just trying their luck with the PBX to get free calls?


INVITE sip:000390237920793@185.43.31.233:5060 SIP/2.0
Via: SIP/2.0/UDP 62.210.103.146:30460;branch=z9hG4bKd7b61754-tv5ODRF2M7Cq1wYx;rport
Contact: <sip:85001@62.210.103.146:30460>
From: <sip:85001@62.210.103.146:30460>;tag=FVLvdl4kCLx8
To: <sip:000390237920793@185.43.31.233:5060>
Call-ID: V4OBFXDLKDQ4.990800.G9PZO5SWEU8P
CSeq: 1 INVITE
User-Agent: IPPhone SP-R50 50.164.5.168 0C11050908A8
Max-Forwards: 70
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Content-Type: application/sdp
Content-Length: 441

Full updated log of last test call with the resulting "not implemented" issue.

.=======================================================.| _____ __ - Pastebin.com

Pastebin.com is the number one paste tool since 2002. Pastebin is a website where you can store text online for a set period of time.

pastebin.com

Thanks

Jon

 

[DigitalDaz]

That invite looks OK to me, you would need to ask AQL what is wrong with it.

Do they not give you any kind of interop instructions?

 

[jddixon]

That invite looks OK to me, you would need to ask AQL what is wrong with it.

Do they not give you any kind of interop instructions?

Hi again

Thanks for that. I did ask for some config guides from AQL but they replied saying there wasn't any! I just got basic config details out of them such as what IPs to whitelist etc, I will log a fault with them and see what response I get.

Thanks for your help

Jon

 

[jddixon]

Hi again

Just heard back from AQL after logging a fault. Turns out the "Error 415 - Unsupported media type" is being caused by 2 instances of a=rtpmap:0 PCMU/8000 in the SDP? Is there any way to correct this easily?