Hi all,
What is Bufferbloat? http://gettys.wordpress.com/bufferbloat-faq/
Maybe this kind of discussion will bring out the John Todds of this
world, I can only hope and dream:
Bufferbloat: http://www.voipusersconference.org/2011/bufferbloat/
Call in and talk to Jim Gettys, who co-developed X
On Thu, 27 Jan 2011 08:46:11 +0100, Magnus Persson mag...@westel.se
wrote:
If you want someting really light weight there is always the old
winpopup protocoll.
Thanks for the tip. It's a nice alternative, although I'd like an app
that keeps a list of pop-ups, in case the user was away and would
On Wed, 26 Jan 2011 14:52:59 +0100, Gilles codecompl...@free.fr
wrote:
Are there open-source solutions you could recommend?
I had another idea: It'd be cool if the application could either just
display CID information, or also search Outlook for a matching Contact
and open the relevant page so
On Thu, Jan 27, 2011 at 11:32:03AM +0100, Gilles wrote:
On Thu, 27 Jan 2011 08:46:11 +0100, Magnus Persson mag...@westel.se
wrote:
If you want someting really light weight there is always the old
winpopup protocoll.
Thanks for the tip. It's a nice alternative, although I'd like an app
On Thu, 27 Jan 2011 12:49:06 +0200, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
An instant-messaging client, as suggested before.
Right. Just a reply to Magnus' suggestion.
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On Thursday 27 Jan 2011, Gilles wrote:
I had another idea: It'd be cool if the application could either just
display CID information, or also search Outlook for a matching Contact
and open the relevant page so that the user can review/add information
for that person. Poor man's CRM :-)
.
http://code.google.com/p/outcall/ ?
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http://www.asterisk.org/hello
On Thu, 27 Jan 2011 11:35:05 +, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
You would do much better in the long run to look at replacing Outlook with
some Open Source alternative -- and sooner, rather than later.
But then, Outlook is pretty much what every office worker uses.
Looks
Not to be redundant, but
http://code.google.com/p/outcall/
On Thu, Jan 27, 2011 at 6:23 AM, Gilles codecompl...@free.fr wrote:
On Thu, 27 Jan 2011 11:35:05 +, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
You would do much better in the long run to look at replacing Outlook with
Hi All,
I would like to implement a call-back option when called user is busy.
Consider this scenario:
1. A caller is calling a number which is busy on another call.
2. The system will prompt the caller (press 3 to be called back etc.) to be
called back when called user is available.
3. Caller
On Thu, 27 Jan 2011 06:24:53 -0600, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
http://code.google.com/p/outcall/
Thanks a lot. I'll check it out.
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Look into Call Completion Supplementary Services for Asterisk 1.8
https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+%28CCSS%29
On Thu, Jan 27, 2011 at 6:48 AM, Harel Cohen ha...@easycall.gi wrote:
Hi All,
I would like to implement a call-back option when called
Kevin
That is grate. I dove into the code and tried to add it my self I added
a F option but I have not figured out the right spot to force the
exclusion to shut off the T38.
Where will the patch be posted?
http://svnview.digium.com/svn/asterisk?view=revrev=304342
Kevin
I tried
On 01/27/2011 09:21 AM, Bryant Zimmerman wrote:
Kevin
That is grate. I dove into the code and tried to add it my self I added
a F option but I have not figured out the right spot to force the
exclusion to shut off the T38.
Where will the patch be posted?
From: Kevin P. Fleming kpflem...@digium.com
Sent: Thursday, January 27, 2011 10:31 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/27/2011 09:21 AM, Bryant Zimmerman wrote:
Kevin
That is grate. I dove into the code
[Jan 27 10:59:56] ERROR[3382]: chan_dahdi.c:12405 dahdi_pri_error:
Should have only transmitted 0 frames!
[Jan 27 10:59:56] ERROR[3382]: chan_dahdi.c:12405 dahdi_pri_error:
Should have only transmitted 0 frames!
I just saw it fly across my CLI.
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HI ,
Please give idea for Multi tenant with Trixbox or elastix.
Thanks
Amardeep Rana
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HI ,
Please give idea for Multi tenant with Trixbox or elastix.
Thanks
Amardeep Rana
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Hi,
Is there any way to have queue member interface answer automatically?
Basically when agentA is called, his phone picks up with no intervention
from his part? (assuming of course he's available and not on the phone, and
not paused).
I already manage this with the Page application (using
I believe all you need to do is to do the same thing just before
running the Queue command...checking
On Thu, Jan 27, 2011 at 10:45 AM, Mike l...@net-wall.com wrote:
Hi,
Is there any way to have queue member interface answer automatically?
Basically when agentA is called, his phone picks up
We do something similar to this by logging a Local channel (eg:
Local/1234@AgentContext) into the queue that passes each call through a few
lines of dialplan code before going to the SIP extension.
Jordan
From: asterisk-users-boun...@lists.digium.com
Ah, there we go, what you'll need to do is some magic with Local
channelscheck out FreePBX's code, it's a little more than I wish
to copy/paste
On Thu, Jan 27, 2011 at 10:55 AM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
I believe all you need to do is to do the same thing just
On 11-01-27 11:41 AM, Amardeep Rana wrote:
Please give idea for Multi tenant with Trixbox or elastix.
http://astbook.asteriskdocs.org
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com http://asterisk.org
Is there any way to have queue member interface answer automatically?
Basically when agentA is called, his phone picks up with no
intervention from his part? (assuming of course hes available and not
on the phone, and not paused).
I already manage this with the Page application
Yeah, if you want per agent, you'll need to use local channels for the
agent interface definition, then in the callagent context, you'll
need to parse the agent's extension and determine if that agent is
supposed to have autoanswer or not... func_odbc and a little dialplan
logic should work nicely
Oh man, I'm sorry, but I laughed so hard at that response, I think I
peed a little :P
To the original poster, Mr Belanger is most definitely being VERY kind
compared to what some people might have responded with
A little effort (and showing that you have put in that effort) goes a
long way
Yeah, if you want per agent, you'll need to use local channels for the
agent interface definition, then in the callagent context, you'll need
to parse the agent's extension and determine if that agent is supposed to
have autoanswer or not... func_odbc and a little dialplan logic should
work
Hi All,
My appologies for the off-topic post, but I thought it would still be
of interest to this list.
If you've been reading our site at http://vuc.me you've no doubt seen
that we have the video call with LifeSize scheduled for Feb 4th. What?
You didn't know? Here's t
scoop:
On 01/27/2011 09:21 AM, Bryant Zimmerman wrote:
Kevin
That is grate. I dove into the code and tried to add it my self I added
a F option but I have not figured out the right spot to force the
exclusion to shut off the T38.
Where will the patch be posted?
From: Kevin P. Fleming kpflem...@digium.com
Sent: Thursday, January 27, 2011 3:08 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/27/2011 09:21 AM, Bryant Zimmerman wrote:
Kevin
That is grate. I dove into the code
Hi all,
Does anyone have any good/bad comments on the A1200P 12-port fxo/fxs card
from OpenVox?
I'll be using one to with 8-12 fxo interfaces. The cards will be plugging
into a cable-modem / phone adapter. We weren't able to port the numbers, so
we're going to use the existing PSTN connection
Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk stop
working after the upgrade. Here is the sip debug:
---
--- SIP read from 208.65.xxx.xxx:5060 ---
INVITE sip:1778xxx@10.11.22.77:5060 SIP/2.0
Via:
On Thu, 27 Jan 2011 14:52:06 -0800
Jian Gao jian@sjgeophysics.com wrote:
Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk
stop working after the upgrade. Here is the sip debug:
---
--- SIP read from
Hey guys,
I'm using asterisk 1.6.2.13 and have an endpoint which uses silence suppression
which I can't turn off. I've set rtpkeepalive=10 in sip.conf [general], as well
as under the peer details for our sip provider but it doesn't seem to do
anything. Rtp debug shows that we are receiving RTP
Hi Mike,
I have used the A1200P without hardware echo cancelation and didn't have any
major issues. The one problem I had was that caller ID simply would not work on
the A1200P, it was fine on the A400P however. This was a year ago though so
things may have changed a little.
Regards,
Ryan.
So, I've done some more testing and got some more info.
I have one endpoint that does silence suppression and one that doesn't. When
the silence suppressing endpoint stops sending RTP, asterisk stops sending RTP
to the other endpoint. I have disabled directmedia and directrtpsetup and it
made
Hi!
I am playing with SendFAX but cant really figure out how it is working.
I have a “fax” /var/spool/asterisk/tmp/fax.tiff that i would like to send to a
“physical” fax at numer 0317998901.
Can some1 write me a simple dialplan that just “grab” the file and send it to
0317998901?
/Magnus--
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