The t, much like reinvite = no keeps asterisk listening to the audio stream to detect dtmf input if dtmf mode is in-band, what is happening is that the sip reinvite is failing, due to a firewall rule or a routing problem and you end up with only one connected RTP stream. Asterisk does not "require" the t option.
Anthony Moe Navid wrote: > Thanks Tony for you reply. > > Do you have any idea why Asterisk require "t" in Dial command? > > Cheers, > > Moe > > On Sun, May 18, 2008 at 1:14 AM, Tony Mountifield > <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: > > In article > <[EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]>>, > Mohammad A. Navid <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> > wrote: > > > > I'm implementing a simple calling card feature for testing > purpose. I have a > > DID number, when I called my DID number and enter the phone > number to call, > > Asterisk would dial the number for me but the sound was only one > way. > > After hours of struggling with the problem, I found out that I > need to add > > "t" to my dial options, this is the correct way of dialing out: > > > > -> Dial(SIP/carrier/3105555555|20|t) > > > > Now I need to know what was going on? Why with option "t" both > parties can > > hear each other, but without option "t" in dial cmd only one > party could > > hear? > > > > Another interesting issue is, if I use Answer() command at the > begining the > > sound becomes one way even if I use "t" in options. > > Try adding "reinvite=no" to the sip.conf or users.conf definition > for your > SIP service provider. > > Cheers > Tony > -- > Tony Mountifield > Work: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> - > http://www.softins.co.uk > Play: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> - > http://tony.mountifield.org > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users