[asterisk-users] packet2packet bridging

[Sameer Rathod]

so In this case If I disable ice support

ie commented the icesuppot=yes from all files

then also I am getting this output


-- Executing [1061 at sameer:1] Dial("SIP/1060-0000008e", "SIP/1061") in new
stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/1061
    -- SIP/1061-0000008f is ringing
    -- SIP/1061-0000008f answered SIP/1060-0000008e
    -- Channel SIP/1061-0000008f joined 'simple_bridge' basic-bridge
<3c12ca41-e180-4fc1-80cf-1339b96da42b>
    -- Channel SIP/1060-0000008e joined 'simple_bridge' basic-bridge
<3c12ca41-e180-4fc1-80cf-1339b96da42b>
       > Bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b: switching from
simple_bridge technology to native_rtp
       > 0x7f6800039020 -- Probation passed - setting RTP source address to
192.168.1.176:8000
       > 0x7f6780045810 -- Probation passed - setting RTP source address to
192.168.1.191:8000







On Wed, Jul 2, 2014 at 8:13 PM, Joshua Colp <jcolp at digium.com> wrote:

> Sameer Rathod wrote:
>
>> yes I had configured
>>
>> icesupport=yes ;
>>
>>
> Asterisk does not support direct media establishment (with either chan_sip
> or chan_pjsip) if secure media (SRTP) or ICE is in use.
>
>
> --
> 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
>



-- 
Regards
Sameer Rathod
8109413462
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140702/c57073ab/attachment.html>