1 Jul
2005
1 Jul
'05
9:16 p.m.
Hi,
On Friday 01 July 2005 13:34, Iqbal wrote:
Not really sure what I want to do with it as yet :-),
but was looking
into the grandstream 102 phones, and they can send DTMF via the INFO
method, which I think helps with call transfers, and then that got me
how does SIP INFO helps with call transfer? If I would program a UA (which I
actually do), I would never submit any tones, no matter if in-band audio,
RFC2833 RTP or SIP INFO, to the remote side while the user enters a transfer
target.
Nils
thinking that if it could do this, then surely I could
use the DTMF to
make ser carry out a few more tasks, that initially I was going to pass
into asterisk and let it do. Not sure what the downside is, except what
you mentioned.
Iqbal
Zeus Ng wrote:
Iqbal,
SIP is supposed to be extensible. So long as the end point understand the
method, it is allowed. A proxy should relay method it does not understand.
The INFO method is defined in RFC and so it's allowed.
Seems like you are trying to extend SER beyond a proxy server. As long as
you have the right module, the thing you are asking is doable. However, I
strongly against this. How would you feel if the DTMF was supposed to be
received by the bank application and somehow SER intercept it and use it
for call routing?
Zeus
Iqbal wrote:
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers is this a "allowed" method if so has
anyone used it to do
clever things with dtmf digit handling, things like executing
a script to amend call forwarding rules etc.
.