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...