Tuesday, March 11, 2008

OCS 2007 and Asterisk integration

Spent some time over the last few days trying to put together an OCS 2007 lab environment with Asterisk integrations and it wasn't exactly smooth sailing.  I have it mostly working now, but there are still some issues with RTP which I hope to resolve when I have some spare time in the coming weeks.

For my setup I deployed OCS 2007 in 2 VMs, one for OCS itself and another for the mediation server.  I used asterisk 1.6-beta5 as it supports TCP SIP and it seems to be working ok.  I can dial out through Asterisk using my home ITSP (PennyTel) through both Communicator 2007 and my Linksys SPA942 IP phone.  I can also receive calls from my mobile to both Communicator and SPA942.

I found a bug in Asterisk 1.6-beta5 where it was not handling the SIP Invite from OCS mediation server properly.  It was causing RTP packets not to be sent from Asterisk to OCS at the start of a call and I traced it back to Asterisk not parsing the "Content-Type" properly.  OCS sends extra data in the "Content-Type" field ";encoding=utf8" which Asterisk was not parsing.  That was an easy fix, one good thing with Open Source software when you actually have time to debug and fix it yourself.

So things seems to be working *mostly* but I still sometime misses out on voice at the beginning of some calls.  I suspect there are still bugs in Asterisk, will post back here for reference if I have to to investigate further.


Anonymous said...

Hi Eddie, am also trying to integrate OCS and Asterisk but my lack of knowledge of Asterisk give me a hard time. Can you explain how you manage to send the Line URI without the + in front of it? Can you give an example of your configuration? I will greatly appreciate.

Anonymous said...

Hi Eddie,

I'm working on the same thing - but I do not have the Linux / Asterisk skills to fix the Content-Type error.

Can you share how you changed that - or better still - a fixed file?

If possiple - a copy of your sip.conf & extensions.conf would also be appreciated.

Thank you in advance for any help...


padams [at] freightliner [dot] bc [dot] ca

Ahmad said...

Hi Eddie,

I was trying to catch you. I have integration problems between OCS and asterisk. please use the following: ahammoud [at] gmai...

we can have a deal about it
waiting for your reply

Nolan said...

Did you ever resolve the issue of missing some audio at the beginning of some calls? I'm struggling with this same issue - but only with certain trunks.


Eddie Tse said...

I gave up after hacking asterisk source code for a while. The delay I was getting with jumping through multiple software with the media stream was too much overhead.

Gilberto Verástegui said...

I have the same issue, any resolution?