Hi Eric,
I am behind nat Is there any solution for the same. My goal is to deduct the balance for the call but free my asterisk server from audio packet load. On Tue, Jul 8, 2014 at 7:51 PM, Eric Wieling <[email protected]> wrote: > I think you will find that direct audio between two endpoints does not > work when NAT is involved. > > > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Sameer Rathod > *Sent:* Tuesday, July 08, 2014 11:18 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] packet2packet bridging > > > > Hi Joshua, > > I had disabled > > ice support and remover encryption= yes > > Then also it is showing the same native_rtp in log > > Could you help me in bypassing asterisk server for audio? > > please help me I am struggling with it form a long time. > > > > > > On Wed, Jul 2, 2014 at 8:21 PM, Sameer Rathod <[email protected]> > wrote: > > -- Channel SIP/1060-0000008e left 'native_rtp' basic-bridge > <3c12ca41-e180-4fc1-80cf-1339b96da42b> > -- Channel SIP/1061-0000008f left 'native_rtp' basic-bridge > <3c12ca41-e180-4fc1-80cf-1339b96da42b> > == Spawn extension (sameer, 1061, 1) exited non-zero on > 'SIP/1060-0000008e' > > here are more generated when I cut the call > > > > On Wed, Jul 2, 2014 at 8:19 PM, Sameer Rathod <[email protected]> > wrote: > > 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@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 <[email protected]> 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 > > > > > > > -- > > Regards > > Sameer Rathod > > 8109413462 > > > > > > > -- > > Regards > > Sameer Rathod > > 8109413462 > > > > -- > _____________________________________________________________________ > -- 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
-- _____________________________________________________________________ -- 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
