Hi,
this java video softphone claims it can operate with Windows messenger. It's
also mentioned on this web page
http://www.voip-info.org/wiki/view/SIP+COMMUNICATOR
But I couldn't find any more info on how to set it up with Asterisk and how
compatible is with other video softphones...
Dear All,
I was trying to limit the number of calls between different located sites in
order to avoid congestion of the bandwidth, but as I found from the mails and
testing that it is easy to do it for the incoming calls by the setgroup() and
group_count while it is the outgoing is hard to track
I'm trying to debug the old call file redial bug
I prepared a call file and trying to setup a call from my remote asterisk
server to my home number. However whenever I dump a call file to
/var/spool/asterisk/outgoing it is just deleted without *any* action
Nothing in the logs, nothing on the
On Sun, 2005-10-23 at 11:42 +0200, Mohamed A. Gombolaty wrote:
Dear All,
I was trying to limit the number of calls between different located sites in
order to avoid congestion of the bandwidth, but as I found from the mails and
testing that it is easy to do it for the incoming calls by the
hi
can we install astbill under mysql 4, or is mysql 5 a must
regards
kanishka
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On Sun, 2005-10-23 at 15:57 +0600, Kanishka Somaratne wrote:
hi
can we install astbill under mysql 4, or is mysql 5 a must
it uses stored procedures which arent available under 4. if you can
work around that you might be able to use 4.x
--
Trixter http://www.0xdecafbad.com Bret McDanel
Hi Trixter,
Yes i did try to make setgroup for the outbound but the problem is after
you move it to the desired context or extension in the gotoif statement
the group that you have set it in is back to zero so I really can't use
it for the outbound, the group used for the outbound will not give
Check if you have Silence Suppression disabled on PSTN line of spa-3000 (admin/advansed/PSTN line).
On Sat, 2005-10-22 at 18:05 -0400, Mike Bernson wrote:
I have asterisk running with sipura 3000 connect to PSTN and
sipura 2000 connected to phones.
On inbound calls I am getting what sounds
When get sip respond 6xx ( such as 603 decline), I want asterisk to
play a voice file to the caller, how to do this in extensions ?
for example, when get 603 respond, play decline.gsm to caller
when get 604 respond, play doesnot-exit.gsm to caller
when
Asterisk guy wrote:
When get sip respond 6xx ( such as 603 decline), I want asterisk to
play a voice file to the caller, how to do this in extensions ?
for example, when get 603 respond, play decline.gsm to caller
when get 604 respond, play doesnot-exit.gsm to caller
Jay Milk wrote:
If that's dishnetwork and they keep charging you their $5 programming
access fee or whatever they call it, just plug it in and confirm that
you get a dial-tone. Then call tech-support and have them adjust
billing -- all they check is that the receiver gets a dial-tone and they
Not an answer to your questions, but just in case you don't know there
is a lot of info on the wiki:
http://www.voip-info.org/wiki/view/AreskiCC+CallingCard+Application
We use Areskicc here, and it works great. However we do not use sip/iax
friends, perhaps both of your problems lie there?
Venerable Sir,
when i start the asterisk server a error message show
that is
chan_oss.c:287 sound thread read error on sound device
resource temperorily unavailable
and hmmming sounds comes
sir what is the problem does my sound card inbuilt in motherboard does not
Has anyone been able to get the 3104 to register more than one line
correctly? It seems to work OK for the first line, but as soon as I
turn on more than one it appears that only the last one is actually
registering corectly. The 3104 sometimes indicates the line is
registered, but * says
Hi!
I have this setup:
Analog phone - Audiocodes MP-114 - Asterisk 1- Aastra 480i
|
\/
Asterisk 2
The codec is alaw on all the calls. Asterisk is CVS-HEAD checked it a couple of hours ago. Asterisk 1 and 2 is connected with a SIP connection using INFO.
The Aastra 480i does not support DTMF
I spent more than 3 weeks, with some little help of people
that belongs to this forum, and after try differents combinations of versions
this is my conclusion:
I tried RH9, FC4 FC4 64
I tried with CVS 1.0.2, and Stable 1.0.9
I tried with spandsp 0.0.2pre18, 0.0.2pre20 0.0.2pre21
Haven't had any issues here with slackware 10.x
On Sunday 23 October 2005 09:23 am, Carlos Alperin wrote:
I spent more than 3 weeks, with some little help of people that belongs to
this forum, and after try differents combinations of versions this is my
conclusion:
I tried RH9, FC4 FC4
2.6.12
On Fri, 2005-10-21 at 09:27 -0700, [EMAIL PROTECTED] wrote:
I received some postings back, as I was trying to do the same thing.
it' is a problem with Kernel 2.6... 2.4 works fine .. this is the
summary
I got from reading the posts before.
I hope that helps... I dont have the
Tressler, Joshua A wrote:
I did a quick Google search of the lists and I hope that I am not
asking a question that has already been answered recently.
I have been working on a interface to use with our CRM software. I am
using the manager interface and mysql to store the changes. The only
Carlos Alperin wrote:
I spent more than 3 weeks, with some little help of people that
belongs to this forum, and after try differents combinations of
versions this is my conclusion:
Please feel free to send every kind of disappointments opinions. That
is going to feel me much better that
On Tue, Oct 18, 2005 at 09:14:31PM -0500, Kevin Scott exclaimed:
As a spin off of that, 10 or so numbers you can call anytime, and then 10
more numbers after that in 24 hours for the random occurrences of 'ordering
pizza'.
But you're right, there are really normally only 10 people I ever try
Hi!
I have this setup:
Analog phone - Audiocodes MP-114 - Asterisk 1- Aastra 480i
|
\/
Asterisk 2
The codec is alaw on all the calls. Asterisk is CVS-HEAD checked it a couple of hours ago. Asterisk 1 and 2 is connected with a SIP connection using INFO.
The Aastra 480i does not support DTMF
Saul,
What you are suggesting follows along the lines of what I am currently
trying however I have determined that if the incoming call has no
callerid, then the channel name is just Zap/1-1/ . For some reason
asterisk doesn't even add the - to the end of the channel name
My concern is
Tressler, Joshua A wrote:
Saul,
What you are suggesting follows along the lines of what I am currently
trying however I have determined that if the incoming call has no
callerid, then the channel name is just Zap/1-1/ . For some reason
asterisk doesn't even add the - to the end of the
I just installed several 3104s in S. Calif. Didn't have any problems--I
was able to call from one line to another on the same unit and between
lines on different units.
Jerry Jones wrote:
Has anyone been able to get the 3104 to register more than one line
correctly? It seems to work OK
Sebastian Milioto wrote:
Hi all,
I have a public ip in Linksys RT31 (2 FXS port + 3 swtich port + 1
uplink port). I want to add behind it, a Linksys pap2 (uplink port + 2
FXS port) with private ip.
I understand that I have to configure Port forwarding or port
triggering (really I'm not sure
Followup, I set a -2.0 gain from my asterisk t1 pbx, and echo seems
mostly gone.
A note, I also turned on the aggressive suppressor in zconfig.h
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory
Wiktor - ADCom Corp.
Sent: Saturday, October
On Sunday 23 October 2005 16:20, [EMAIL PROTECTED] wrote:
Followup, I set a -2.0 gain from my asterisk t1 pbx, and echo seems
mostly gone.
A note, I also turned on the aggressive suppressor in zconfig.h
It's the turning on the agressive mode that did it. Agressive mode works by
turning the
I have just a quick setup question about how some of you
have hardware setup.
Basically, for a system that has an average volumes of
calls in an office setting, are you using one or two
network cards. I am just wondering if it owuld be any
advantage to having one NIC for the extensions and
Sorry guys I forgot to mention that in my setup I always enable
agressive in zconfig
On 10/23/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Sunday 23 October 2005 16:20, [EMAIL PROTECTED] wrote:
Followup, I set a -2.0 gain from my asterisk t1 pbx, and echo seems
mostly gone.
A note,
The CEO of SIPMEDIA lives down the road from me. I will find out
what's going on, and report back.
What I do know is that it's a real company and not one running behind
a lemonade stand, backed by major players in the industry. I can of
course not give out more than this on the structure of the
On Sunday 23 October 2005 18:02, C F wrote:
Sorry guys I forgot to mention that in my setup I always enable
agressive in zconfig
Yuck. I find the agressive echo canceller totally unacceptable.
-A.
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I've stumbled upon a very interesting phenomenon.
In setting up a trunk from iConnectHere (ich) I mistakenly input
type=from-pstn
type=friend
This, in [EMAIL PROTECTED] using AMP.
No User Context entries are made
and a DID entry (with the full 11 digit number) is in the DID settings.
I now
Why?
On 10/23/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Sunday 23 October 2005 18:02, C F wrote:
Sorry guys I forgot to mention that in my setup I always enable
agressive in zconfig
Yuck. I find the agressive echo canceller totally unacceptable.
-A.
When I try to create sip/iax friend from web interface it says
Could not open buddy file '/etc/asterisk/additional_areskicc_sip.conf'
I tried creating the file manually without luck.
Make sure the user your web-server runs as can write to and read from the file.
Second I am unable to
Silence Supp Enable is No.
Sergey Okhapkin wrote:
Check if you have Silence Suppression disabled on PSTN line of spa-3000
(admin/advansed/PSTN line).
On Sat, 2005-10-22 at 18:05 -0400, Mike Bernson wrote:
I have asterisk running with sipura 3000 connect to PSTN and
sipura
Hi guys,
Just a quick question. I need to buy a dual T1 card and I'm debating
between TE210P or the Sangoma A102u. Any recommendations?
Thanks,
Waldo
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On Sunday 23 October 2005 18:30, C F wrote:
Why?
Because it sounds like ass. I (and my customers) are used to the full-duplex
nature of the telephone system. Half duplex sounds very unnatural.
-A.
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Well, I don't think that that's what I hear when I enable it. It works
really nice for me.
On 10/23/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Sunday 23 October 2005 18:30, C F wrote:
Why?
Because it sounds like ass. I (and my customers) are used to the full-duplex
nature of the
On Sunday 23 October 2005 19:59, C F wrote:
Well, I don't think that that's what I hear when I enable it. It works
really nice for me.
Chacon son gout. :-) if you are listening to someone and you start to talk
(to interrupt say) their voice disappears immediately. Or if there's
background
Hello,
Watch out there, that's a very touchy issue. I'll try to lay out the
technical and non-technical points of view.
First the purely technical point of view:
The Digium TE210P/TE205P is basically the TE4XXP(quad card) with only
two ports included instead of four. It uses almost exactly the
Your answer was in queues.conf that's why you only got 1 reply.
Where in queues.conf? Could you please point out where? Thanks
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Matt,
Thanks for the reply.
I had read your blog entry a few days ago and that's when I started
doubting whether or not to go with Sangoma.
I've only heard good things of Sangoma. However, I only have
experience with TE410 and because of my limited experience with
Asterisk, I was a bit
KRTorio wrote:
Where in queues.conf? Could you please point out where? Thanks
Check /usr/src/asterisk/configs/queues.conf.sample if you have updated.
Now to state the obvious:
; Calls may be recorded using Asterisk's monitor resource
; This can be enabled from within the Queue application,
Waldo Rubinstein wrote:
The only thing I wished was that the Digium cards worked in 3.3V and 5V
motherboards without having to specify which one you are going to
deploy it on. I got somewhat screwed on the TE410P because of that
reason :(
The warranty issue is a big difference. Why
On Sunday 23 October 2005 21:40, Kevin Bockman wrote:
I agree on both points. I'm not sure if anyone from Digium actually
reads the -users lists though.
Kevin Fleming slogs through this list just as I do. It's a lot to keep up
with. Hell I think he even does it off the clock, as I do.
I
Hello there,
I'm having problems with Festival text-to-speech generator. Apparently,
Asterisk connects to the Festival server, but no audio is generated.
Does anybody know:
a) if Asterisk is compatible with Festival 1.4.2?
b) it is possible to download new voices for text2wave (for the
How about any IAX softphones for the pocket pc platform?
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Same question that before:
Mandrake/Mandrive 2.6.13.4 (thanks for the info)
What version of Asterisk?
What version of Spandsp?
What version of Libtiff?
What version of Libtiff-devel?
And the million dollars question: Is the fax working? (Lets say more than
50% of the cases?)
Thanks for your
Good for you
Slackware 10.x (exactly 10.?) to be more accurate?
What version of Asterisk?
What version of Spandsp?
What version of Libtiff?
What version of Libtiff-devel?
And the million dollars question: Is the fax working? (Lets say more than
50% of the cases?)
Thanks for your info. I'm trying
I don't know if this noise is related to our noise.Note: forwarded message attached.---BeginMessage---
Silence Supp Enable is No.
Sergey Okhapkin wrote:
Check if you have Silence Suppression disabled on PSTN line of spa-3000
(admin/advansed/PSTN line).
On Sat, 2005-10-22 at 18:05
By the way,
After follow all the rules,
RH9 with Libtiff 3.5.7, Libtiff-devel 3.5.7, OpenSSL-Devel, Readline41,
Ncurses4, Ncurses C++ Devel, SOX, Asterisk 1.0.9 Stable, Spandsp 0.0.2pre21,
App_txfax.c app_rxfax.c dated October 21, 2005. This time everything was
smoth and nice.
Sunday 23:30 the
On 10/20/05, Darrick Hartman [EMAIL PROTECTED] wrote:
Leif Madsen wrote:
PS: If the Asterisk Documentation Project website becomes slow due to
the number of people accessing it at once, we appoligize and
appreciate your patience. For those of you who are able to obtain the
full copy,
I use the command asterisk -RT to connect to a running asterisk box.
There must be some changes to the latest CVS upgrade:
1. it does not remember anything anymore what I have done in the
previous connection. I could reconnect to the asterisk box and with
arrow up I could see all my
On Sun, 2005-10-23 at 23:57 -0400, Leif Madsen wrote:
On 10/20/05, Darrick Hartman [EMAIL PROTECTED] wrote:
Is there any reason why the book wasn't released as a single pdf rather
than the individual chapter pdf's? Using pdftk, I merged the pdfs back
into a single document (11mb), then
On 10/24/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
On Sun, 2005-10-23 at 23:57 -0400, Leif Madsen wrote:
On 10/20/05, Darrick Hartman [EMAIL PROTECTED] wrote:
Is there any reason why the book wasn't released as a single pdf rather
than the individual chapter pdf's? Using
I've been using spandsp 0.0.2pre18 and 0.0.2pre21 with libtiff 3.7.3, Asterisk
1.0.7 (now 1.0.9) and the Linux kernel 2.6.11.5 kernel with results that are
good enough for me (I'm using fax over IP with the G.711 codec).
On Sunday 23 October 2005 13:23, Carlos Alperin wrote:
I spent more than
On Mon, 2005-10-24 at 00:18 -0400, Leif Madsen wrote:
The problem here is that you CAN'T make derivitive works. I think you
have the wrong license (however I can't verify since I can't get to
the Creative Commons website right now). There is no derivitive works
allowed of the book (and no
On 10/24/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
I dont know who has access to post blog entries at creativecommons.org
but Glenn Otis Brown wrote in support of repackaging saying it helps the
original author (in his example songwriters).
On Mon, 2005-10-24 at 01:00 -0400, Leif Madsen wrote:
However, I was basically just erring on the side of caution --
re-distribution of the works is permitted (this is why we are allowed
to have so many mirrors!). I would assume repackaging of the work
would be considered part of the
Regarding the pdfs, I seem to be missing some pages between the
COPYRIGHT.pdf (page iv) and foreword.pdf (page ix)..
Is this one of those blank page intentionally inserted here moments
or is something really dropped off here?
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On 10/24/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
And this should be why I am allowed to create a single PDF of the file
to be distributed to our mirrors (which btw, I have done, so once the
mirrors are updated, you will be able to get the file in a single PDF
as opposed to
On 10/24/05, Wai-Sun Chia [EMAIL PROTECTED] wrote:
Regarding the pdfs, I seem to be missing some pages between the
COPYRIGHT.pdf (page iv) and foreword.pdf (page ix)..
Is this one of those blank page intentionally inserted here moments
or is something really dropped off here?
Hrmm...
On 10/23/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
don't know if there is much difference in maintaining a Sangoma
install vs a Digium install. I do know the setup may be somewhat
different, but that's a one time deal (sometimes).
There really isn't much difference once you've gotten it
On 10/23/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Sunday 23 October 2005 21:40, Kevin Bockman wrote:
I agree on both points. I'm not sure if anyone from Digium actually
reads the -users lists though.
Kevin Fleming slogs through this list just as I do. It's a lot to keep up
with.
On 10/24/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
And this should be why I am allowed to create a single PDF of the file
to be distributed to our mirrors (which btw, I have done, so once the
mirrors are updated, you will be able to get the file in a single PDF
as opposed to
On Mon, Oct 24, 2005 at 12:04:40PM +0800, Ronald Wiplinger wrote:
I use the command asterisk -RT to connect to a running asterisk box.
There must be some changes to the latest CVS upgrade:
1. it does not remember anything anymore what I have done in the
previous connection.
Why
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