Joel,   Why do you have SIP GTD enabled on your Cisco router?  FreeSWITCH won't do anything with this information anyway, you might as well disable it and multipart handling in general.  It's probably enabled by default because you're using QSIG (I'm not sure) but this doc should be able to help: http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/sip_cg-isd...   Or you could try to tweak multipart handling in FS using this variable: http://wiki.freeswitch.org/wiki/Variable_sip_copy_multipart Also - why do you have compact headers turned on? On Tue, Dec 28, 2010 at 8:35 PM, Joel Mange <telconettech@yahoo.com> wrote:

Hello.  New to FS and in need of help.

Environment is as follows:

Linksys SPA-962 (x94140) <---> FS <---sip---> Cisco3845 <---qsig---> TDM PBX <--> TDM phone (x91237)

Can complete call from TDM phone to Linksys SPA-962.  Can't complete call from Linksys to TDM phone:  TDM phone rings, but call fails when it goes off-hook.  Getting the following sofia error:  nua(0x8b60cf0): event i_media_error 988 Incomplete offer/answer.

I think the above message tells me that the other side of the call sent an invalid SDP.  Having trouble identifying the invalid SDP and making necessary correction.

Siptrace and router running-config at:  http://pastebin.freeswitch.org/14896

Any help appreciated.

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