SIP Trunk to Asterisk - Unsupported sip incoming method on trunk interface <REGISTER> - ignori

[g18c]

Hi guys, i have setup (or trying to!) a SIP trunk between an IP Office v6.0 (1803) and Asterisk.

I have a trunk which works one way (IP Office to Asterisk), but Asterisk to IP Office i am getting the following error:

SIP Reg/Opt Rx: 17
REGISTER sip:192.168.10.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.60.5:5060;branch=z9hG4bK363ee040
Max-Forwards: 70
From: <sip:lon-pbx-1@192.168.10.5>;tag=as7fa01211
To: <sip:lon-pbx-1@192.168.10.5>
Call-ID: 3df506e02f2b24fe296246df1d531dd7@[::1]
CSeq: 778 REGISTER
User-Agent: FPBX-12.0.76.2(11.19.0)
Expires: 120
Contact: <sip:s@192.168.60.5:5060>
Content-Length: 0

Sip: Unsupported sip incoming method on trunk interface <REGISTER> - ignoring
Sip: Cannot accept call: reason=405 (Method Not Allowed)

Any ideas please?

 

[amriddle01]

It shouldn't be set to register, you can't register SIP trunks to an IP Office they need to be direct connect

 

[pairofeyes]

We have a similar problem with a client R7.0(36) that is interfacing with a 3rd party IVR/VRU using SIP trunks. Works fine to get into IVR from a short code or DID via SIP trunk, and interrogate database, etc., but when IVR tries to transfer back to the phone system extension [to a human] it drops the call.

Monitor and Wireshark reveal that the main error is:

405 Method not allowed

Once we also see this error:

SIP/2.0 481 Dialog/Transaction Does Not Exist

Error after blind transfer

The URI /SIP trunks, CODEC’s are set up as follows:
REFER checked, incoming and outgoing set to 'always'
call route method: Request URI
SIP/URI Via: none , regist: none [local UR, contact, Disp name all set to * wildcard]
codec lockdown: unchecked ; all 4 codec's checked

We have tried many combinations on advice from tech support areas, but they say ‘coding looks good, it should work’ and AVAYA level IV says ‘hey why don’t you try upgrading to 9.0 or 9.1, see if that works’ ..Well we would do that first thing, but this is a 24/7 IT tech support environment and bringing down the system [and major issue unknowns going R7->R8->R9] on the ‘chance’ it might cure the problem, and no guarantees... The client is understandably hesitant, but can be done.

Any suggestions welcome! thanks

 

[quazimotto]

There is a method in FAQ for IPO to Asterisk. Would that work?

Quaz

 

[pairofeyes]

I will look, thanks