Hi,
I know it is a bit off-topic, but I'd like to ask the community what is the
current most supported way to deal with DTMF?
I'm looking for an all-SIP system and I'm mostly interested in the end
devices support of the different methods (DTMF in-band audio, DTMF RTP
telephony events packets, SIP
re
or install new apps.
Thanks a bunch!
Cesc
-- Forwarded message --
From: Cesc Santa <[EMAIL PROTECTED]>
Date: Fri, Nov 28, 2008 at 3:26 PM
Subject: Asterisk RTP pager
To: [EMAIL PROTECTED]
Hi,
I came across your "RTPpage" application and just made me very happy.
If
Hi,
I am trying to use an Avaya 4602 phone, which I just updated from a
very old SIP software to the latest I could find on avaya's site
(032207). The upgrade went fine and it gets registered on the Asterisk
server.
Now, a couple of glitches, though.
- The phone's web server is not working ... so
On 8/10/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>
> On Fri, Aug 10, 2007 at 11:35:32AM +0200, Cesc Santa wrote:
> > Hi,
> >
> > I have asterisk 1.2.18.
> > I just took a peak at the command: > show translation
> > and I saw that I can only con
inline
On 8/10/07, Gordon Henderson <[EMAIL PROTECTED]> wrote:
>
> On Fri, 10 Aug 2007, Cesc Santa wrote:
>
> >
> > I can use them ... but cannot do translation ... only "pass-through" ...
> it
> > complains that
> > there is no function to
inline ...
On 8/10/07, Gordon Henderson <[EMAIL PROTECTED]> wrote:
>
> On Fri, 10 Aug 2007, Cesc Santa wrote:
>
> > Hi,
> >
> > I have asterisk 1.2.18.
>
> Installed from binary or compiled by yourself?
I compiled it myself ...
> I just took a peak at
Hi,
I have asterisk 1.2.18.
I just took a peak at the command: > show translation
and I saw that I can only convert from/to ulaw, ulaw, gsm and slin.
No speex, no ilbc ... do I need a license or compile something extra?
The G723, 726 and 729 ... I need a license, is that it? one for all of them?
o
Hi,
I have asterisk 1.2.18.
I am trying to get asterisk to react to an (out of dialog) REFER ...
see below. I get a 603 (no dialog) ... and in the code (sip.conf:3277)
a comment being able but not supporting it??
Any pointers would be great ... is it a configuration option?
REFER sip:[EMAIL PROT
Hi,
I need an Asterisk with G729 support. Preference is with Asterisk
1.2(.18), but if not possible, then it can be 1.4.
Question is, can I enable G729 for both protocols? do the H323
implementation allow it? I found the codec support for H323 in 1.2.18
very poor ... only got u/a-law to work ... n
