Hi George,
I tried the nightly build and also Bria. I can replicate the same issue on both.
This morning I made many successful calls in succession. This evening
it was intermittent again.
Could it be the mobile network is blocking the RTP but it seems odd it
works sometimes and not others.
I am having trouble getting Google Chrome to accept a WebRTC call coming from
Asterisk, even though Firefox can (now) accept the same call without issue.
My setup is as follows:
Server:
CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146
asterisk-11.21.0 patched to work around
El 20/01/16 a las 16:25, Alex Villacís Lasso escribió:
I am having trouble getting Google Chrome to accept a WebRTC call coming from
Asterisk, even though Firefox can (now) accept the same call without issue.
My setup is as follows:
Server:
CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4
El 20/01/16 a las 18:33, Alex Villacís Lasso escribió:
El 20/01/16 a las 16:25, Alex Villacís Lasso escribió:
Partial fix: Google Chrome accepts the call if videosupport is set to "no".
This is the SDP of the successful INVITE that Chrome accepts:
INVITE
Hello List;
I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and I
am getting the following debug, can someone advise me about the solution:
<--- SIP read from Provider_IP_Address:5083 --->INVITE
sip:22021782@Asterisk_IP_Address:5060 SIP/2.0 Via: SIP/2.0/UDP
Hello;
Thanks a lot for your kindly reply.Actually the alaw is enabled at asterisk but
what I got to know from the other side that they only enabled ulaw. Below is my
asterisk sip configuration for the sip trunk. Please advise.
On Wednesday 20 Jan 2016, bilal ghayyad wrote:
> Hello List;
> I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and
> I am getting the following debug, can someone advise me about the
> solution: <--- SIP read from Provider_IP_Address:5083 --->INVITE
> . [stuff deleted]
