Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week.

Thanks for any help.

Ronald Wiplinger wrote:

Tiago Stein D`Agostini wrote:

Hi,

Ie been looking for some time how to use asterisk to initiate SIP connections between 2 IP phones, but afetr initiated the communication making the RTP go directly from one telephone to the other, without passing by asterisk. Unfortunately I found no explanations of how to do it.

Does anyone care to give a pointer to any explanation about how to do it?

canreinvite=yes
and look at the options for dial()


Thanks in advance



bye

Ronald Wiplinger
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to