Hi Jared & Kevin,
Thanks for taking the time to answer my questions. I wonder if I could just
be reading the tcpdump incorrectly? I'm still seeing rtp streams (and
Jared, I have modified the dial string to remove the L)..
Here's a screenshot of what I'm seeing in wireshark. I really appreciate
th
A2billing usually stays in the media path due to the dialstring
parameters that it uses to cut a call off when the balance would reach
$0. To get Asterisk to step out of the media path, I had to change
dialcommand_param_sipiax_friend and dialcommand_param to |60|S(14400)
which lets all calls go to
On 05/21/2012 03:45 PM, David Wessell wrote:
More specific on sip.conf
In sip.conf I have a trunk specified for the SIP provider, and a trunk
specified for the PBX itself.
Do I need to specify directmedia=yes on both sides?
Yes, it has to be set on both peers involved in the bridged call.
-
More specific on sip.conf
In sip.conf I have a trunk specified for the SIP provider, and a trunk
specified for the PBX itself.
Do I need to specify directmedia=yes on both sides?
Thanks for your help. I'll be testing it in a few minutes.
Thanks
David
On Mon, May 21, 2012 at 2:18 PM, Kevin P.
On 05/21/2012 12:54 PM, David Wessell wrote:
So I need directmedia set in sip.conf on the LCR trunk.
1) Do I need it in the individual trunk settings for each pbx? Or is
in sip.conf enough?
You say 'in sip.conf' multiple times, but that's far too vague to mean
anything. sip.conf is a configur
So I need directmedia set in sip.conf on the LCR trunk.
1) Do I need it in the individual trunk settings for each pbx? Or is
in sip.conf enough?
2) Do I need anything on the pbx side that we are hoping to transfer media to?
3) How long into the call before the media is transferred over?
Thanks
Da
On 05/21/2012 11:46 AM, David Wessell wrote:
Hi Kevin,
Thank you. Here's the requested information.
1) The Trunk is running 1.6.2.9. Also it's running a2billing.
2) The PBX is running asterisk 1.8.12.0 along with FreePBX.
3) I did directmedia on the trunk and canreinvite on the pbx since
they w
Hi Kevin,
Thank you. Here's the requested information.
1) The Trunk is running 1.6.2.9. Also it's running a2billing.
2) The PBX is running asterisk 1.8.12.0 along with FreePBX.
3) I did directmedia on the trunk and canreinvite on the pbx since
they were different versions.
Thansk
David
On Mon,
On 05/21/2012 07:03 AM, David Wessell wrote:
I am attempting to get an asterisk server to step out of the media
path, but am running into a brick wall. Can someone assist? Here's my
setup..
Ultimate SIP Provider ---> LCR Trunk (Asterisk 1.6) > PBX (Asterisk 1.8).
In order to be able to
All g711 calls, and the only nat is on the endpoint.
Snom M9 Phone (behind nat) -> PBX (Public IP) -> LCR Trunk (Public IP)
-> SIP Provider (Public IP)...
I'm expecting the LCR trunk to get out of the media path and connect
the PBX with the SIP Provider
Thanks
David
On Mon, May 21, 2012 at
Hi,
Can you check if there is any transcoding involved with these calls, or
maybe some NAT handling needs to be done by asterisk so it's not stepping
out of the media-path !?
Regards,
Sammy
On Mon, May 21, 2012 at 5:03 PM, David Wessell wrote:
> I am attempting to get an asterisk server to ste
I am attempting to get an asterisk server to step out of the media
path, but am running into a brick wall. Can someone assist? Here's my
setup..
Ultimate SIP Provider ---> LCR Trunk (Asterisk 1.6) > PBX (Asterisk 1.8).
I am attempting to get the trunk to step out of the media stream.
There i
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