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 <[email protected]> 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 <[email protected]> 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 <[email protected]> > 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: [email protected] | SIP: [email protected] | 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 > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > _____________________________________________________________________ > > -- 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 > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > -- > www.ringfree.biz > 828-575-0030 > -- -- www.ringfree.biz 828-575-0030
-- _____________________________________________________________________ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
