NO JOY Trunks will not register if more than one provider used

[SMTC]

I had an extremely stable although deprecated PIAF installation. Within the last 60 days my trunks stopped working correctly in that they would not register if all of them were enabled.

I have trunks from:
Comwave
Freephoneline
Unlimitel (Primus) x2

If you disable all but one provider, they register (including the two discrete ones fro Unlimitel). BUT you do have to reboot the PBX to get one up. Logs look pretty much the same as:

http://pbxinaflash.com/community/threads/lost-trunk-registration.17133/#post-110672

Registrations get sent, but time out.

I have since:
a) replaced the PIAF with an up to date new RaspPi2 running Incredible PBX
b) I have swapped out my DD-WRT E3000 router with the backup WRT-54GL which I knew was flawless before although slow for in-house wireless clients.
c) I have exchanged my ROGERS DOCSIS Modem with a newer one.

I have been and am using NAT and have the settings in Asterisk SIP settings for External address set up correctly as was before.

Here's a funny rub: If you put a WRONG address in the Asterisk SIP settings, the trunks will all register. Of course the downside of this is that all external phones have no audio so that is not a work-around.

I have been "working on the PBX" now since before Christmas and am at a total loss.

SIP SET DEBUG ON yields nothing particularly interesting.

Providers while helpful only prove their trunk is fine with a test on a Softphone client.

The Google Voice trunk, new with the Incredible PBX installation, works fine this whole time.

Anyone finding that its maybe something to do with SANDVINE's Traffic Policy Engine that ROGERS (and BELL) apprently use?

https://www.sandvine.com/technology/policy-engine.html

Anything else I can try?

 

[SMTC]

And I've also tried putting the PBX in the DMZ and even turned off IPTABLES temporarily. No difference.

 

[SMTC]

Excerpt from DEBUG


---
[2016-01-31 20:21:08] NOTICE[3819]: chan_sip.c:15347 sip_reg_timeout: -- Regi stration for '[email protected]' timed out, trying again (Attempt #1 55)
Really destroying SIP dialog '[email protected]' Method : REGISTER
Retransmitting #1 (NAT) to 209.217.98.154:5060:
REGISTER sip:sip05.unlimitel.ca SIP/2.0
Via: SIP/2.0/UDP 99.231.XXX.XXX:5060;branch=z9hG4bK3c33d9a1;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as6fe35140
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 256 REGISTER
Supported: replaces, timer
User-Agent: LinksysPAP(2T)
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0

 

[SMTC]

No ideas here guys? I think I'm going to try a FreePBX install to see what happens...

 

[wa4zlw]

hi there...did you put your voip gateways into iptables to allow them? I have multiple gateways running without any issues. If you're not using one of the predefined TRUNKs then you need to add in your gateways IP(s) otherwise you're not going to work.

Leon