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
