[asterisk-users] simultaneous use of chan_sip/chan_pjsip

2015-08-13 Thread Marek Červenka
hello, is it possible simultaneously use chan_sip and chan_pjsip? if yes, can you recommend settings i'm thinking about - chan_sip - for sip hardphones/softphones (sip udp 5060) - chan_pjsip - for webrtc -- --- Marek Cervenka

[asterisk-users] One way audio - doesn't seem to be NAT issue

2015-08-13 Thread Stefan Viljoen
Hi D'arcy Have you checked your RTP port ranges (I'm sure you have), and also that the server IP for RTP as specified in the initial SIP is correct? Not sure how this will relate to your setup, but we had something similar here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP

Re: [asterisk-users] simultaneous use of chan_sip/chan_pjsip

2015-08-13 Thread Rusty Newton
On Thu, Aug 13, 2015 at 3:54 AM, Marek Červenka cerv...@fpf.slu.cz wrote: hello, is it possible simultaneously use chan_sip and chan_pjsip? if yes, can you recommend settings i'm thinking about - chan_sip - for sip hardphones/softphones (sip udp 5060) - chan_pjsip - for webrtc You can

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue

2015-08-13 Thread D'Arcy J.M. Cain
On Thu, 13 Aug 2015 10:41:31 +0200 Stefan Viljoen viljo...@verishare.co.za wrote: Have you checked your RTP port ranges (I'm sure you have), and also Yes. The ATA is using a range well within the range open on the server. that the server IP for RTP as specified in the initial SIP is correct?

Re: [asterisk-users] simultaneous use of chan_sip/chan_pjsip

2015-08-13 Thread Marek Červenka
Dne 13.8.2015 v 17:20 Rusty Newton napsal(a): On Thu, Aug 13, 2015 at 3:54 AM, Marek Červenka cerv...@fpf.slu.cz mailto:cerv...@fpf.slu.cz wrote: hello, is it possible simultaneously use chan_sip and chan_pjsip? if yes, can you recommend settings i'm thinking about -

[asterisk-users] Is peer order in sip.conf important?

2015-08-13 Thread Murthy Gandikota
Hi All Noticed in sip.conf that the asterisk (v11) is sensitive to the order of peers.  Here  is my sip.conf [general] context = demo  ;              Default context for incoming calls bindport = 5060  ;              UDP Port to bind to (SIP standard port is 5060) bindaddr = 0.0.0.0  ;