SIP TRUNK 1

[veenstrahiem]

I have programmed a SIP trunk on a 3300 mcd4.2.
The strange thing is, that I can use the sip-trunk on outgoing calls just fine, but I cannot get the incoming calls to work.
The call gets a disconnected tone as if the number does not excist.
Up to now the siptrunk was mainly used for outgoing calls, with a separate ISDN trunk for incoming calls, but we really want to use the incoming on the siptrunk as well.

commands:

SIP REGISTRAR CONFIG

Registrar Configuration
-----------------------
Domain: mitel.bv.dns IP Address 192.168.20.2
Allow Third Party Registrations: no
Default Expiry Duration: 3600 seconds
Maximum Configurable Expiry Duration: 7200 seconds


SIP REGISTRAR CONTACTS ALL

Registrar Entry(s)
--------------------


SIP REGISTRAR STATS

Registrar Statistics
--------------------
Current: 0
Accepted: 0
Refreshed: 0
Total Failed: 415406
(Rejected 33)
(400 Bad Request 1024)
(403 Forbidden 0)
(403 Forbidden Blocked 0)
(404 Not Found 414349) see Maintenance logs
(423 Min Expires 0)
Registration request timeout: 0
CallControl response timeout: 0


This seems to be a problem, a lot of 404 not found calls, but wat can I do to solve this?

 

[danramirez]

what do you get from sip provider? do a CCS TRACE ENABLE CONT to see what digits you are receiving, then handle the digits in your trunk attributes.

or... program the trunk as non dial in to make incoming calls are arriving to your Mitel

 

[wireman50]

Have you programmed the absorb digits in trunk attributes and iether set the extension numbers to match the last digits or set up a system speed call to match the last digits and pointed to an extension?

 

[DrDeee]

Have you programmed SIP Peer Profile Assignment by Incoming DID?

 

[fissure]

you need to list the numbers (that the calls are coming into) in:
SIp Peer Profile Assignment by Incoming DID
or the Mitel responds with 404 (not found)
the numbers you list here are the full numbers (before any digits stripped in 'trunk attributes'

in case of queries, use maintenance commands
sip all set level 3
sip all set storage c
sip all trace on

then dial in, and look for the INVITE message
then do:
sip all trace off



look for something like:


2011-12-08 20:37:56 SSP Network->STS Level = 3 Unique Id = 0
INVITE sip:xxx@172.16.35.5 SIP/2.0
Via: SIP/2.0/UDP 85.92.88.91:5060;branch=z9hG4bK76678aef;rport
From: "xxx" <sip:xxx@xxx>;tag=as664cef26
To: <sip:AAAAAAAAAAAAA@172.16.35.5>
Contact: <sip:xxx@xxx>
Call-ID: 2abdd1b8672d8656453054e0375e0c1f@xxx
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 08 Dec 2011 20:40:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 262

whatever shows in TO, replaced with AAAAAAAAAAAAA, above, is what you need in 'SIp Peer Profile Assignment by Incoming DID'


in (virtual) MCD5 'sip all trace on' became:
sip all tail on
I think

 

[veenstrahiem]

thanks all, I will try.

 

[veenstrahiem]

I finally had the time to test these, and thanks, I solved it.
After starting the sip traces, I saw that an extra digits was send by the SIP-trunk provider. I inserted this digit, and the numbers came ringing on my phones,

THANKS.