http://pastebin.freeswitch.org/22575

Here is the Siptrace


On Fri, May 16, 2014 at 9:16 AM, Joel White <joelewhite@gmail.com> wrote:
inserting <action application="set" data="sip_copy_multipart=false"/> in the dialplan allows calls to ring the Polycom phone on Kamailio

However no media is exchanged or at least no audio is heard on either end


On Fri, May 16, 2014 at 8:53 AM, Joel White <joelewhite@gmail.com> wrote:
Thank you

Question for you Kristian

This Multipart SIP variable, is it set in the dialplan or globally?


On Thu, May 15, 2014 at 5:58 PM, Kristian Kielhofner <kris@kriskinc.com> wrote:
I don't miss seeing this ridiculous Nortel SIP one bit...

So:

- The Nortel includes a phone-context attribute in the username portion of the URI.
- The Nortel uses a multipart SIP body (of which the SDP is one part).
- The Nortel uses a Contact address with a domain in the URI and a "maddr" attribute to signify the actual signaling IP.

When forwarding the INVITE to the Polycom (via Kamailio, apparently) more than likely it's blowing up on the multipart SDP. Check this out:


There is no shortage of hoops to jump through with the CS1000.


On Thu, May 15, 2014 at 4:21 PM, Joel White <joelewhite@gmail.com> wrote:
This siptrace is from the CS1000 to Asterisk to FreeSWITCH

This go through successfully and media was exchanged.  This of course being a temporary solution until I can make FreeSWITCH work.


On Thu, May 15, 2014 at 4:18 PM, Joel White <joelewhite@gmail.com> wrote:
This siptrace is the CS1000 calling directly to FreeSWITCH

52321 is an extension on the Nortel


On Thu, May 15, 2014 at 1:54 PM, Brian West <brian@freeswitch.org> wrote:
A sip trace plus log would be helpful, 'sofia global siptrace on’, place the call and paste bin the output to pastebin.freeswitch.org
--
Brian West


Twitter: @FreeSWITCH , @briankwest














On May 15, 2014, at 12:31 PM, Joel White <joelewhite@gmail.com> wrote:

I am still working on integrating Nortel with FreeSWITCH

I have the calls coming in through FreeSWITCH and parsing/matching XML

Here is the latest issue though

the call runs through XML, looks like it is working

then the calls drops and no audio was ever heard on the Nortel extension calling in

I get SIP errors 415 from the Nortel side and a 501 from the FreeSWITCH

The call hangs up with an reason 79

https://wiki.freeswitch.org/wiki/Hangup_Causes

I checked and the audio is PCMU:8000

What am I missing here?


Something to mention, I have been using an Asterisk as a gateway to pass calls from the Nortel to FS

There is no issue when I do it this way, but Asterisk was a temporary patch and I need to make this work on FreeSWITCH

Thank you in advance for any assistance

Joel
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

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--
Kristian Kielhofner

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
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http://wiki.freeswitch.org
http://www.cluecon.com

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