[OpenSIPS-Users] Asterisk Unrecognized sip header

[Travis Manson-Drake]

Hello everyone.

Hope your all doing well!

I seem to be having an issue in which when a call is sent through OpenSIPS to my Asterisk PBX asterisk with eventually send a BYE with a hang up Cause of 111/unrecognized sip header. I looked at the headers of all my packets but can't find anything out of the norm. has anyone experienced this before and ideas on what it might be or what I might check?

I found a few article on asterisk forums mention NAT issues, but I've implemented a NAT helper into my routing logic so that shouldn't be the case.

Thank you all for your time


Travis Manson-Drake
Voice Systems Analyst


Simply Bits, LLC
T: 520.545.0311  F: 520.545.7252
E: travism at simplybits.com<mailto:travism at simplybits.com>
5225 N. Sabino Canyon Road
Tucson, AZ 85750
Support Hotline: 520.545.0333



-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20160331/c3c943b2/attachment-0001.htm>