[asterisk-users] PJSIP issues with handling incoming calls

Joshua Colp jcolp at digium.com
Tue Sep 2 08:08:11 CDT 2014


Nick Awesome wrote:
> register =>  73432260005:pass at 10001
> register =>  73432260050:pass at 10002
>
> [10001]
> type=peer
> host=80.75.132.66
> context=dialmap
> [10002]
> type=peer
> host=80.75.132.66
> context=dialmap

Can you provide a sip debug of calls to both of these? I'm confused how 
that... works...

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org



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