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[manavid at gmail.com]

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.
One more interesting thing, my carrier for calling out only accepts G7.29
where as my DID provider passes calls as ulaw. However, when using
voipjet.com (as secondary carrier) which carries out the calls as ulaw,
having the Dial cmd without "t" option works fine.

I'd appreciate if someone explain to me the following questions:
1) Why when there is codec difference, Dial cmd needs the "t" option
2) Why using Answer cmd causes problem in this case (all the cases, when
using same codec and also different codecs)
3) Why with same codecs, Dial cmd does not need "t" option?

Thanks

Moe
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[anthonyf at rockynet.com]

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: