Well, can we see the requests being sent to Asterisk? :)
On 03/31/2016 01:48 PM, Travis Manson-Drake wrote:
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:trav...@simplybits.com <mailto:trav...@simplybits.com> 5225 N. Sabino Canyon Road Tucson, AZ 85750 Support Hotline: 520.545.0333 _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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