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
--
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Marek Cervenka
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
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
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?
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
-
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 ;