On 21:07, Wed 03 Oct 07, Atis Lezdins wrote:
> I clearly understand that it's that way. However my question is about
> usability - if i do `asterisk -rx "show anything"` - i would really want only
> that output, and wouldn't want asterisk to transfer anything else trough
> remote socket. What i
On Wednesday 03 October 2007 20:45:09 Jason Parker wrote:
> Atis Lezdins wrote:
> > Hi,
> >
> > Guys, could you please take a look at
> > http://bugs.digium.com/view.php?id=10847
> >
> > The problem is that i expect `asterisk -rx "show queues"` to return only
> > result of "show queues", as it woul
Atis Lezdins wrote:
> Hi,
>
> Guys, could you please take a look at http://bugs.digium.com/view.php?id=10847
>
> The problem is that i expect `asterisk -rx "show queues"` to return only
> result of "show queues", as it would be in CLI with verbosity/debug 0.
> However, sometimes i get full CLI
3 okt 2007 kl. 18.58 skrev Power, Paul C.:
> I got this working on 1.2.9 and 1.4.11.
>
> We use Polycom phones with and OpenSER proxy and AudioCodes gateway to
> the PSTN world.
> Progressinband is set to never.
>
>
Again, that's early media, not early bridging. Early media support
has been in
I got this working on 1.2.9 and 1.4.11.
We use Polycom phones with and OpenSER proxy and AudioCodes gateway to
the PSTN world.
Progressinband is set to never.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Chris Ziomkowski
> Sent: Wednesday,
Hi,
chan_ss7.so is loaded. I get incoming calls from the PSTN. Chan_ss7 CLI
commands as you can see when typing "help ss7" work. I was also able to
watch the signalling links going up on the CLI after asterisk has started.
I get reload-not-supported message while trying "module reload chan_ss7.so
Hi,
Guys, could you please take a look at http://bugs.digium.com/view.php?id=10847
The problem is that i expect `asterisk -rx "show queues"` to return only
result of "show queues", as it would be in CLI with verbosity/debug 0.
However, sometimes i get full CLI output for quite long period of ti
Caio Begotti wrote:
> On Wed, 03 Oct 2007 07:05:59 -0500
> "Kevin P. Fleming" <[EMAIL PROTECTED]> wrote:
>
>> It is in fact quite nice :-) I'm sure there will be official
>> pictures posted on the web site once we have moved in
>
> This picture still applies or has the building changed? :-)
> htt
>
> It works well both way for * 1.4.2 but as I tried it with 1.4.4 I got
> problem making outbound calls. See below
>
>
>
> -- Executing [EMAIL PROTECTED]:3] Dial("SIP/192.168.178.251-081ffab0",
> "ss7/siuc/6974223663") in new stack
>
> [Oct 2 18:59:38] WARNING[2539]: channel.c:3099 ast_request
Hi.
I've patched Safira's chan_ss7 for use with * 1.4. I've compiled the
chan_ss7.so outside asterisk installation path and then copied it to
/usr/asterisk/lib/modules
It works well both way for * 1.4.2 but as I tried it with 1.4.4 I got
problem making outbound calls. See below
-- Exe
On Wed, 03 Oct 2007 07:05:59 -0500
"Kevin P. Fleming" <[EMAIL PROTECTED]> wrote:
> It is in fact quite nice :-) I'm sure there will be official
> pictures posted on the web site once we have moved in
This picture still applies or has the building changed? :-)
http://www.russellbryant.net/gallery2
Brian Capouch wrote:
> Inquiring minds wonder whether there might be a picture or two around
> that could be perused. . . I hear it's pretty nice.
It is in fact quite nice :-) I'm sure there will be official pictures
posted on the web site once we have moved in, but as of today the
parking lot w
Thank you John! I was sitting in front of you at the dev summit at
astricon and brought this subject up. This patch is MOST welcome. I
will need to get it merged into my asterisk. Then I will test and keep
you posted..
Doug
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL P
Andrew,
Do you mind sharing how you got it working in branch 1.4?
Thanks!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, October 01, 2007 6:21 PM
To: asterisk-dev@lists.digium.com
Subject: [asterisk-dev] Thanks for the G.7
3 okt 2007 kl. 10.48 skrev Sergio Garcia:
>
>
> Thanks for the clarification.
> Probably I'm again wrong, but I assume that the main problem is
> knowing
> if the remote peer has compatible codecs in order to keep asterisk
> in the
> media loop or not. Could it be solved if we implement the o
Thanks for the clarification.
Probably I'm again wrong, but I assume that the main problem is knowing
if the remote peer has compatible codecs in order to keep asterisk in the
media loop or not. Could it be solved if we implement the option of
sending INVITEs without SDP and sending it in the AC
The early bridge is an asterisk concept where we set up the call
directly
between two endpoints instead of making the decision later and sending
re-invites.
We do support early media but not with PRACK. That's a different thing.
/O
3 okt 2007 kl. 09.57 skrev Sergio Garcia:
>
>
>
> -- O
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