John Lange wrote:
Specifically, callers trying to record their voicemail greetings will
have jittered audio.
Setting jbenable=yes in zaptel.conf in order to dejitter sip audio is
very confusing and I know there will be a _lot_ of people wondering
about this besides me.
I strongly agree wit
I want to create a CLI or Manager command to busy and un-busy channels
in a PRI trunk, without actually making real calls.
Does chan_zap have any API functions that I can call to do this, or do
I need to create some new functionality? How about libpri?
The objective is that a busied-out channel (
On Tuesday 24 October 2006 11:08, John Lange wrote:
> jbenable=yes in sip.conf should tell Asterisk to dejitter incoming audio
> FROM SIP channels when the receiving leg can not handle dejitter on its
> own.
> One simple example demonstrates why this makes sense; in the case where
> the sip channe
On Tue, Oct 24, 2006 at 07:28:57AM -0500, Kevin P. Fleming wrote:
> Whatever we do, we will not offer formats to the second endpoint that
> were not chosen during negotiation with the first endpoint. We don't
> want the call to be setup using direct media and a format which we
> cannot support shou
Thanks for the clarification Slav. I appreciate the work you've done on
this. A decent JB is very important.
So, the bottom line is you set jbenable=yes on channels you want to send
dejittered audio too. In most cases this would be any channel which can
not handle its own dejitter such as Zap chan
If you want to listen into the Developers Summit meeting at Astricon you
can call 972-961-7666 or IAX2/[EMAIL PROTECTED]/4569.
Jeremy McNamara
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I've a question. Does asterisk use threading when execute application? I
mean whether all applications are
duplicated for each user in memory (which are registered as
ast_register_application). And how many users can asterisk supply at
one time usually?
Asterisk use one thread for each channel ex
Hi,
I’m posting here as I have found an issue in
1.4, and hoping someone might be able to help.
I have setup a call queue in asterisk, a call comes
into the queue, asterisk calls the agents, an agent answers the call fine, but
if they try and transfer the call, asterisk drops out wi
On Tue, 2006-10-24 at 07:25 -0500, Kevin P. Fleming wrote:
> Luigi Rizzo wrote:
> > yes there is no fopencookie manpage, but the call is present e.g. on
> > the linux box where you gave me an account, and where i actually
> > tested that the code was correctly handling https.
> > You can try for yo
Brian Candler wrote:
> 3. Revert to 1.2 behaviour (i.e. offer all codecs with media proxying, and
> re-INVITE to set up native bridging afterwards). However, re-order the codec
> list so that the codecs included in the incoming SIP INVITE appear before
> the others. This might make it a bit more li
Luigi Rizzo wrote:
> yes there is no fopencookie manpage, but the call is present e.g. on
> the linux box where you gave me an account, and where i actually
> tested that the code was correctly handling https.
> You can try for yourself.
>
> Note that fopencookie(), as well as funopen(), has no re
In article <[EMAIL PROTECTED]>,
Yuan Qin <[EMAIL PROTECTED]> wrote:
>
> Hi, all:
>
> Asterisk1.2.10, in pbx.c, the ast_custom_function_register is
>
> int ast_custom_function_register(struct ast_custom_function *acf)
> {
> if (!acf)
> return -1;
>
> /* try to lock function
On Mon, Oct 23, 2006 at 05:47:56PM -0500, Kevin P. Fleming wrote:
> > (However, I have no idea what these devices would do if you completed the
> > call, and then they had to use the 'WIBBLE' codec to send audio...)
>
> I don't think that my proposal is practical; instead, what I am going to
> do
Hi, all: Asterisk1.2.10, in pbx.c, the ast_custom_function_register isint ast_custom_function_register(struct ast_custom_function *acf) { if (!acf) return -1; /* try to lock functions list ... */
if (ast_mutex_lock(&acflock)) { ast_log(LOG_ERROR, "Unable to lock functio
On Mon, Oct 23, 2006 at 02:37:39PM +0100, Brian Candler wrote:
> Here's one I knocked together in Perl:
> http://pobox.com/~b.candler/software/testsiperror
>
> The fake codec it offers is "WIBBLE/8000".
>
> And the results I get when pointing it at various SIP UAs:
>
> Audiocodes Tulip ATA (2.
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