Thanks for the response.. I did change it in the [general] settings.My setup is something like I have a remote conference (not meetme) which will send reinvite to redirect the RTP flow to a different server to load balance.There are three clients who join in the conference and i can listen to two other clients speak from the third client but when i record the conversation my recording of one of the clients ends before the stipulated hangup time. I am guessing this is because one of the clients doesn't understand what to do with a reinvite.. Any suggestions.In the SIP.conf i have changed the directmedia option to no and also enabled the ignoresdpversion option.
On Tue, Jul 3, 2012 at 10:01 PM, SamyGo <[email protected]> wrote: > I don't think you can set SIP properties in some variables anywhere in > asterisk dialplan or call file. What you can do is change the directmedia > options of the SIP or any other channel you're using. i.e if your call file > has > > CHANNEL=SIP/12345@latestgateway > > Then change the properties of the [latestgateway] in sip.conf. Also if > you're using an IP address directly > > CHANNEL=SIP/[email protected] > > Then you can change the directmedia directive in sip.conf [general] > settings. > > Hope it helped. > > BR > Sammy Go. > > On Wed, Jul 4, 2012 at 2:08 AM, sathiish kumar > <[email protected]>wrote: > >> I am using call files to make calls to a remote machine but can't seem >> to quite understand the directmedia options that are set by default in >> Asterisk.Is there any way i can specify the directmedia options using call >> files? >> >> >> >> -- >> _____________________________________________________________________ >> -- 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 > -- -------------------------------------------------------------------- Sathiish Kumar Mohan Kumar Technical Operations Intern, Citrix Online 7408 Hollister Avenue,Goleta, Santa Barbara CA-93106 ---------------------------------------------------------------------
-- _____________________________________________________________________ -- 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
