Configuring SIP Trunks on Trixbox

[topanaris]

Ok now i have gotten the trunks to work matronyx does your caller ID feature work? As is do you see the numbers for persons dialing in? If yours does work could you share your config.

[bogle]

Does anyone have the inbound calling set up?

and is there a way to actual have the sip peer display a OK status when sip show peers is executed?

[bogle]

Its ok I figured it out,

the Incoming user call should have been named bizvoip.cwjamaica.com and did not include qualify within the trunk settings to verify status.

[matronyx]

Yo on a real, I just checking back this post. Sorry for not answering your question topanaris.
Any questions regarding this just shout me on my cell 395-9168.
I hardly get the time to check the forums as much as I'd like to but id be happy to share the knowledge on here.

Next thing im working on is DIGICEL's SIP config. They refuse to go public IP with their package so its WIMAX private network for now. Ive gotten the registration and all down but for some reason getting a "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21" error. Will link them about it in the morning.

[matronyx]

UPDATE
===============

Digicel SIP trunk now working
Took a while, a few nights with the digicel techs analyzing traffic on both ends etc, but in the end it work, inbound and outbound calls.

Here are a few points gathered from this setup:
1) CODECS
Ensure youre using the correct codec on your asterisk box AND handsets. For this setup g729a was the standard codec.
To check your phone handset for the codec its using do a "sip show peer [extension]" where extension is that particular handset.
Note the order for the codecs. On my grandstream phones I had to implicitly set each codec option to g729a to force it to handshake properly for outbound calls.

2) ROUTES
Ensure that your routes to digicel's backend (their switch, IDU blah blah blah) is setup correctly on your asterisk box.

3) USE WIRESHARK!!!!
Analyze your packets to help with troubleshooting authentication problems.
Without this little tool, this project would be a fail.

Here is all the trunk settings that should get you up and running, free of cost of course

PEER:
username=[yournumber]
fromuser=[yournumber]
type=peer
secret=[yourpass]
port=5060
fromdomain=digiceljamaica.net
host=[yourhost]
defaultexpiry=3600
dtmfmode=rfc2833
canreinvite=no
insecure=very
disallow=all
allow=g729
sendrpid=yes&pai
nat=yes

USERCONTEXT:
[yournumber]

REGISTRATION string:
[yournumber]:[yourpass]@digiceljamaica.net/[yournumber]~3600

[Sal]

any one having problems with their LIME sip trunks? mine have stopped registering.....nothing changed on the PBX