[asterisk-users] Asterisk and the media path

[David Wessell]

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
the suggestions.

Screenshot:
http://dl.dropbox.com/u/4156401/Screenshot%20from%202012-05-23%2007%3A39%3A51.png


pcap: http://dl.dropbox.com/u/4156401/trace3000.pcap

Thanks
David

On Wed, May 23, 2012 at 7:41 AM, David Wessell <david at ringfree.biz> wrote:

> 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
> the suggestions.
>
> Thanks
> David
>
>
>
>
> On Mon, May 21, 2012 at 6:08 PM, Jared Geiger <jared at compuwizz.net> wrote:
> > 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 14400 seconds. The default uses the L
> > parameter. You need to use the S parameter instead. However the S
> > parameter doesn't like very large numbers in Asterisk 1.4 so I've just
> > hard set mine.
> >
> > ~Jared
> >
> > On Mon, May 21, 2012 at 5:18 PM, Kevin P. Fleming <kpfleming at digium.com>
> wrote:
> >> 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.
> >>
> >>
> >> --
> >> Kevin P. Fleming
> >> Digium, Inc. | Director of Software Technologies
> >> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype:
> kpfleming
> >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> >> Check us out at www.digium.com & www.asterisk.org
> >>
> >> --
> >> _____________________________________________________________________
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >> New to Asterisk? Join us for a live introductory webinar every Thurs:
> >>              http://www.asterisk.org/hello
> >>
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> >
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> >               http://www.asterisk.org/hello
> >
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> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> --
> www.ringfree.biz
> 828-575-0030
>



-- 
--
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828-575-0030
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