[asterisk-users] One way sound when Using Dial cmd without "t" option (SOLVED) Need explanation

[Anthony Francis]

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 
> <tony at softins.clara.co.uk <mailto:tony at softins.clara.co.uk>> wrote:
>
>     In article
>     <28749f210805170447w7e2da378vb11d12bdf8dd4b81 at mail.gmail.com
>     <mailto:28749f210805170447w7e2da378vb11d12bdf8dd4b81 at mail.gmail.com>>,
>     Mohammad A. Navid <manavid at gmail.com <mailto:manavid at gmail.com>>
>     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: tony at softins.co.uk <mailto:tony at softins.co.uk> -
>     http://www.softins.co.uk
>     Play: tony at mountifield.org <mailto:tony at mountifield.org> -
>     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