Sameer Rathod wrote:
= Using SIP RTP CoS mark 5
-- Executing [1061@sameer:1] Dial("SIP/1060-00000088", "SIP/1061")
in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/1061
-- SIP/1061-00000089 is ringing
> 0x7f67f90b43c0 -- Probation passed - setting RTP source address to
192.168.1.176:8000 <http://192.168.1.176:8000>
-- SIP/1061-00000089 answered SIP/1060-00000088
-- Channel SIP/1061-00000089 joined 'simple_bridge' basic-bridge
<1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4>
-- Channel SIP/1060-00000088 joined 'simple_bridge' basic-bridge
<1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4>
> Bridge 1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4: switching from
simple_bridge technology to native_rtp
> 0x7f67f90b43c0 -- Probation passed - setting RTP source address to
192.168.1.176:8000 <http://192.168.1.176:8000>
> 0x7f6780047090 -- Probation passed - setting RTP source address to
192.168.1.191:8000 <http://192.168.1.191:8000>
== WebSocket connection from '192.168.1.191:54390
<http://192.168.1.191:54390>' closed
Are either side using encryption or ICE?
--
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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users