Sip mid-call DTMF

Can asterisk send dtmf mid-call?
I have a sip v2 device that uses sip method info to send and receive signaling mid-call. I would like to be able from asterisk send this information to the device. The device is using dtmf 1 and 0 out of band to transmitt contact closure from one device to another but i would like to be able to send this information directly from asterisk. Is this possable? The device is used to control a 2-way radio ptt signal.

Comments or sugestions appreciated…

Yes. The sip.conf option to enable this mode should be fairly obvious.

The question is not how to configure sip method info. But how to send from asterisk to the sip device mid-call (during the call)!

Press the digit keys on the other phone. You can also do third party sending using AMI.


first of all thanks for the replys. I have a sip device that requires a 1 to key a output relay a 0 to unkey that relay. That is easy enough to send form the conected sip phone ( or soft phone ) the problem is this unit also requires that 1, 0, or a 2 to be received at less than every 60 seconds or it times out and drops the connection, this keep alive digit 2 is what i need asterisk to send about every 30 seconds to keep the connection alive for the duration of the confrence call ( conference connection needed so that i can have mulpiple operators connected to the device). After a little more reading looks like i should be able to do this with [applicationmap] although i havent quite figured that out yet. Your thoughts would be very appreciated.

I think you will need to use AMI.