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--
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 n
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.
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
On Thu, 27 Jan 2011 14:52:06 -0800
Jian Gao 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 208.65.xxx.xxx:5060 -
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:
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 a
From: "Kevin P. Fleming"
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 and tried to
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?
http://svnview.digium.com/svn/a
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:
http://www.voipuse
>
> 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
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 in
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 nicel
> > 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
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
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
wrote:
> I believe all you need to do is to do the same thing just before
> running the Queue comm
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
[mailto:asterisk-users-boun...@lists.digium
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 wrote:
> Hi,
>
>
>
> Is there any way to have queue member interface answer automatically?
> Basically when agentA is called, his phone picks up with no int
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
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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[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.
--
___
From: "Kevin P. Fleming"
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 and tried t
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?
http://svnview.digium.com/svn/a
> 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=rev&rev=304342
Kevin
I tried e
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 wrote:
> Hi All,
>
> I would like to implement a call-back option when called user is busy.
>
On Thu, 27 Jan 2011 06:24:53 -0600, Sherwood McGowan
wrote:
>http://code.google.com/p/outcall/
Thanks a lot. I'll check it out.
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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 h
Not to be redundant, but
http://code.google.com/p/outcall/
On Thu, Jan 27, 2011 at 6:23 AM, Gilles wrote:
> On Thu, 27 Jan 2011 11:35:05 +, A J Stiles
> wrote:
>>You would do much better in the long run to look at replacing Outlook with
>>some Open Source alternative -- and sooner, ra
On Thu, 27 Jan 2011 11:35:05 +, A J Stiles
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 like it's possible to access Ou
http://code.google.com/p/outcall/ ?
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http://www.asterisk.org/hello
ast
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 :-)
.
On Thu, 27 Jan 2011 12:49:06 +0200, Tzafrir Cohen
wrote:
>An instant-messaging client, as suggested before.
Right. Just a reply to Magnus' suggestion.
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N
On Thu, Jan 27, 2011 at 11:32:03AM +0100, Gilles wrote:
> On Thu, 27 Jan 2011 08:46:11 +0100, Magnus Persson
> 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
On Wed, 26 Jan 2011 14:52:59 +0100, Gilles
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 that the user can revi
On Thu, 27 Jan 2011 08:46:11 +0100, Magnus Persson
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 like
to see who
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 Wi
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