[OpenSIPS-Users] Asterisk Unrecognized sip header

Alex Balashov abalashov at evaristesys.com
Thu Mar 31 19:49:53 CEST 2016


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:travism at simplybits.com <mailto:travism at simplybits.com>
> 5225 N. Sabino Canyon Road
> Tucson, AZ 85750
> Support Hotline: 520.545.0333
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


-- 
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/



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