search for: pubopensips

Displaying 1 result from an estimated 1 matches for "pubopensips".

2013 Sep 17
1
RTP not being switched between both SIP endpoints
...d so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call coming from opensips. Directmedia is not specifically defined so its using the asterisk default value. * Name : vmpubopensips3 Description : Secret : <Not set> MD5Secret : <Not set> Remote Secret: <Not set> Context : from-pubopensips Record On feature : automon Record Off feature : automon Subscr.Cont. : <Not set> Language : Tonezone : <Not se...