Not that I am aware of. From: asterisk-users-bounces@xxxxxxxxxxxxxxxx [mailto:asterisk-users-bounces@xxxxxxxxxxxxxxxx] On Behalf Of Sameer Rathod Hi Eric, I am behind nat Is there any solution for the same. My goal is to deduct the balance On Tue, Jul 8, 2014 at 7:51 PM, Eric Wieling <EWieling@xxxxxxxxx> wrote: I think you will find that direct audio between two endpoints does not work when NAT is involved. From: asterisk-users-bounces@xxxxxxxxxxxxxxxx [mailto:asterisk-users-bounces@xxxxxxxxxxxxxxxx] On Behalf Of Sameer Rathod 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 <sameer@xxxxxxxxxxxxx> wrote: -- Channel SIP/1060-0000008e left 'native_rtp' basic-bridge <3c12ca41-e180-4fc1-80cf-1339b96da42b> here are more generated when I cut the call On Wed, Jul 2, 2014 at 8:19 PM, Sameer Rathod <sameer@xxxxxxxxxxxxx> 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
-- SIP/1061-0000008f is ringing On Wed, Jul 2, 2014 at 8:13 PM, Joshua Colp <jcolp@xxxxxxxxxx> wrote: Sameer Rathod wrote: yes I had configured Asterisk does not support direct media establishment (with either chan_sip or chan_pjsip) if secure media (SRTP) or ICE is in use.
-- Regards Sameer Rathod 8109413462
Regards Sameer Rathod 8109413462
Regards Sameer Rathod 8109413462
Regards Sameer Rathod 8109413462 |
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