Hi,
Let me provide the details first:
* Asterisk 1.8.32 on CentOS behind the NAT firewall
* Two (2) SIP trunks with "canreinvite=no" and "directmedia=no"
If a call comes from either trunk and is bridged to a local extension
there is never a problem with audio. The same is true for outbound
Hi folks.
I have a couple of questions regarding RTP.
The background of my inquiry is that I have packet captures of SIP and
RTP traffic on an Asterisk and Broadworks SIP trunk and the RTP many
times has a time stamp that rewinds by 480 using g.711u. The Sequence
number continues to increment
Hello;
Have you run the script that's included in the Asterisk distribution
that lists and installs the needed dependencies? It's called
"install_prereq" and it's in the contrib/scripts directory. Hope this helps.
Regards;
John V.
-Original Message-
From:
On Tue, Aug 29, 2017, at 10:57 AM, Thomas wrote:
> Hello,
>
> since al long time I have used UNIQUEID for identify calls in my
> dialplan,
> statistics...
>
> Now I have had an problem, after I have checked log file I found out
> following:
>
> calls same time ( hours:seconds) came in.
>
>
Hello,
since al long time I have used UNIQUEID for identify calls in my dialplan,
statistics...
Now I have had an problem, after I have checked log file I found out following:
calls same time ( hours:seconds) came in.
CallID, DID, channel name (3cf9 to 3cfa) are different.
Only
