It's funny, but though I think nothing of having a linux box as a pbx,
on 24/7 for years, I can't imagine using windows this way. I think
there's little or no market for this whereas if there were a fanless,
diskless embedded solution for just under $200 that came configured
with the account (IAX
On Fri, 14 Dec 2007 17:34:04 -0600, "Michael Graves"
<[EMAIL PROTECTED]> wrote:
>Yes, the market is potentially huge...for a packaged solution.
If all it takes in plugging the PCi card in their PC, and running
setup.exe, it's no worse than installing a printer. I would imagine
that the standard in
bilal,
flash operator panel (fop) or any of the asterisk gui does this...
asteriskNow for example...
http://www.asternic.org/
daveC
bilal ghayyad wrote:
Hi All;
Is there an GUI for Asterisk that can help in showing
the call flow (who is in progress, who is connected,
called number, ...)?
hi,
ya, there is one s/w whiche is freely available for linux os as *
events.tar * .
it is in php. you can use this.
regards,
Bhrugu Mehta
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Try to increase the value of busycount. It may help to solve the problem.
On Dec 14, 2007 10:47 PM, bilal ghayyad <[EMAIL PROTECTED]> wrote:
> Hi All;
>
> For me, I am in Kuwait and using the TDM22B and I used
> all the below settings and did not resolve my problem,
> I do not know if there is an
Yes, was my mistake, just the first pair of bits are changed to 0x0,
that is, if initially is 1001, 0001 is the seize state.
On Dec 14, 2007 1:04 PM, Roger C. Beraldi Martins
<[EMAIL PROTECTED]> wrote:
> Moises,
>
> I was reading about your first reply and you said in the 2nd step:
>
> >
> >2. lib
On Fri, Dec 14, 2007 at 09:58:52AM +0800, Rilawich Ango wrote:
> > >
> > > busydetect = yes
> > > hanguponpolarityswitch = yes
> >
> > Which of the two?
> >
> > busydetect will work almost always. But it is suboptimal: it may sotimes
> > accidentally detect running calls. And it takes a few seconds
On Fri, Dec 14, 2007 at 06:01:49AM +0100, Vincent wrote:
> On Fri, 14 Dec 2007 14:50:28 +1000, [EMAIL PROTECTED] wrote:
> >Erm, there just might be, take a look at this...:
>
> Ah yeah, forgot about $angoma ;-) I'll restate this as: No card for
> home/SOHO use, ie. in the $50-100 range for the sin
On Fri, 14 Dec 2007 20:43:05 +0100, Vincent wrote:
>On Fri, 14 Dec 2007 10:30:46 -0700, [EMAIL PROTECTED] wrote:
>>That said, consider the potential market size for people, the DIY sorts,
>>who would have Asterisk in their homes.
>
>Precisely: The home/SOHO market is huge, and providing an IVR + P
On Thu, 2007-12-13 at 22:21 -0600, Tilghman Lesher wrote:
> On Thursday 13 December 2007 19:55:39 Vincent wrote:
> > I was wondering why there doesn't seem to a Windows version of Zaptel,
> > making the Digium and its clones unavailable for a Windows PBX.
>
> Because nobody has done it yet. The r
On Fri, 2007-12-14 at 20:38 +0100, Vincent wrote:
> On Fri, 14 Dec 2007 10:51:10 -0500, Lee Jenkins <[EMAIL PROTECTED]>
> wrote:
> >I have to reboot my desktop xp box daily for it to run well.
>
> I haven't rebooted my XPSP2 in months,
I also have no problem with the stability of Windows XP
Thanks, I believe that is what I was looking for.
joe a.
>>> On 12/14/2007 at 3:44 PM, "Zaheer K. Master" <[EMAIL PROTECTED]>
wrote:
> Hi Joe,
> In your SIP.conf, under [general] try setting "externip=XXX.XXX.XXX.XXX" to
> your public IP address.
>
> Hope this helps,
> Zaheer
>
> -Original
Ryan M. Colbert wrote:
>We have an issue with Linksys SPA2102-NA ATA's where there is a several second
>delay between when you finish dialing and when it sends the commands on to *.
>Has anyone else seen this before? If so, is there a quick/easy solution?
>
>
Yes, it is quick and easy. Mak
Hi Joe,
In your SIP.conf, under [general] try setting "externip=XXX.XXX.XXX.XXX" to
your public IP address.
Hope this helps,
Zaheer
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Acquisto
Sent: Friday, December 14, 2007 2:44 PM
To: asterisk-users@lis
We have an issue with Linksys SPA2102-NA ATA's where there is a several second
delay between when you finish dialing and when it sends the commands on to *.
Has anyone else seen this before? If so, is there a quick/easy solution?
Ryan M. Colbert
Director of Information Technology
Rissman, Bar
hi vincent,
In the UK you can have multiple pots lines with the same telephone
number. but you would need more fxo lines for this.
Regards
Robb
Vincent Li wrote:
> Hi Lists,
>
> I have one box with two FXO and two FXS ports, it is running asterisk
> inside. I have one sinle POTS line connecte
Hi Lists,
I have one box with two FXO and two FXS ports, it is running asterisk
inside. I have one sinle POTS line connected to the one FXO and two
phone sets connected to the FXS port.
Extension 6003 is asigned to one fxs and 6004 is asigned to another
fxs, the two extensions can call each other
On Friday 14 December 2007 14:43, Vincent wrote:
> OTOH, having to run a separate PC just to handle calls from a single
> POST line AND having to install Linux + Asterisk on this thing... It'd
> have to be an appliance (which I haven't seen avaiable in this price
> range).
Didn't you just define a
On Fri, 14 Dec 2007 10:30:46 -0700, [EMAIL PROTECTED] wrote:
>That said, consider the potential market size for people, the DIY sorts,
>who would have Asterisk in their homes.
Precisely: The home/SOHO market is huge, and providing an IVR + PCI
card combo for Windows for, say, $200, would probably
Vincent wrote:
> On Fri, 14 Dec 2007 10:51:10 -0500, Lee Jenkins <[EMAIL PROTECTED]>
> wrote:
>> I have to reboot my desktop xp box daily for it to run well.
>
> I haven't rebooted my XPSP2 in months, and I let it run 24/7, with a
> bunch of apps open at all times. And this is a 300E no-name box.
Trying to setup SIP to register with a VOIP provider. I am behind a firewall
(IPCOP) with NAT.
Getting this, in CLI with SIP debug on.
Retransmitting #2 (no NAT) to aa.bbb.ccc.ddd:5060:
REGISTER sip:voip-xxx.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.xxx:5060;branch=z9hG4bK727a6144;rport
From: ;tag
On Fri, 14 Dec 2007 15:47:38 +1100, Paul Hales
<[EMAIL PROTECTED]> wrote:
>Umm - you could just buy a SPA-3000/3102/3666/etc.
Thanks but I prefer PCI cards. Less cables, less power units that can
burn, less mess :-)
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On Fri, 14 Dec 2007 10:51:10 -0500, Lee Jenkins <[EMAIL PROTECTED]>
wrote:
>I have to reboot my desktop xp box daily for it to run well.
I haven't rebooted my XPSP2 in months, and I let it run 24/7, with a
bunch of apps open at all times. And this is a 300E no-name box.
If your PC is so unstable,
Roger,
I think you should call your telco support and ask them if they
receive your sizing and if they are sending the seize ack.
On Dec 14, 2007 5:57 AM, Roger C. Beraldi Martins
<[EMAIL PROTECTED]> wrote:
> Dears,
>
> Here is the logs when I put loglevel=255 on unicall.conf, I have use
> max-wa
Moises,
I was reading about your first reply and you said in the 2nd step:
>
>2. libmfcr2 will set the ABCD bits to 0x0 (000) ( normally the ABCD
>bits are in Idle 1001 ). Setting the ABCD bits to 0x0 is our way to
>tell the far end ( the telco ) that we want to start a call, this is
>known as th
bilal ghayyad wrote:
> cd /usr/src/asterisk-1.4/channels/h323
>
> When I type make, it gives me:
> make: Nothing to be done for 'degault'
This is *exactly* what showed up on your session? The word 'degault'
does not appear in the Makefile at all, so if that is the message that
you got then your
Hi All;
Is there an GUI for Asterisk that can help in showing
the call flow (who is in progress, who is connected,
called number, ...)? I was think in AsteriskNow does
this? Any advise?
Regards
Bilal
Be
On 12/14/07, Tony Plack <[EMAIL PROTECTED]> wrote:
>
> > /etc/asterisk/externsions.conf file:
> >
> > --
> > exten => 10100,1,Wait(4)
> > exten => 10100,2,Playback(transfer,noanswer)
> > exten => 10100,3,Dial(${PHONE30},30,t)
> > exten => 10100,4,Background(extension)
> > exten => 10100,5,Backg
I would not be surprised at all if they had enough interest in a shared
premium short code for the same dollar amount, they would implement it
for a cut of the action as I said, and credit by keyword.
I will not post their info to the list since I certainly do not want to
get a company in troub
I'm a platform agnostic. I need to use a bit of everything in my daily
work. I will say that my current XP desktop has been very reliable. It's
not uncommon for it to stay up for a couple of weeks at at time without
a reboot. My linux and FreeBSD systems routinely go months untouched.
That said, c
Hi All;
I am trying now to compile h323 to be able to use it,
I did the pwlib and openh323 successfully and I
exported the PWLIBDIR=/usr/src/pwlib_v1_10_0 and the
OPENH323DIR=/usr/src/openh323_v1_18_0, then I was need
to compile h323 as following:
cd /usr/src/asterisk-1.4/channels/h323
When I ty
> On Thu, 13 Dec 2007, Daniel M. Baker wrote:
>
>> I will be out of the office from Dec. 10-13.
>>
>> Thanks,
>> Dan Baker
>> Ursuline Academy
>> 314-984-2828
>>
> Thanks for the "heads-up." We should be finished removing all of
> your valuables by then :)
>
I thought the same thing
> /etc/asterisk/externsions.conf file:
>
> --
> exten => 10100,1,Wait(4)
> exten => 10100,2,Playback(transfer,noanswer)
> exten => 10100,3,Dial(${PHONE30},30,t)
> exten => 10100,4,Background(extension)
> exten => 10100,5,Background(is-curntly-unavail)
> exten => 10100,6,Voicemail()
> exten
Doug wrote:
> At 19:55 12/13/2007, Vincent wrote:
> >Hello
> >
> >I was wondering why there doesn't seem to a Windows version of Zaptel,
> >making the Digium and its clones unavailable for a Windows PBX.
> >
> >Is the Zaptel/Zapata combo too *nix-centric?
> >
> >Thanks.
>
> Windows is a ha
Hy all,
I have some queues and I want to monitor them.
I want to choose if I am going to monitor all the calls of the queue (via
monitor-format variable of queues.conf) , or only some calls (via wW parameter
of Queue command)
But this last option doesnt work for me. I have:
exten=>30,1,Ans
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tilghman Lesher wrote:
>
> gcc 4.2 has a bad optimization that has yet to be tracked down, which causes
> the transcoding code to be incorrectly built. There are two workarounds for
> this problem: either use gcc 4.1 or enable the DONT_OPTIMIZE comp
[incoming]
exten => 2125551211,1,GoTo(companyA,1)
exten => 2125551212,1,GoTo(companyB,1)
exten => 2125551213,1,GoTo(companyC,1)
[companyA]
exten => 2000,1,Dial()
[companyB]
exten => 2000,1,Dial()
[companyC]
exten => 2000,1,Dial()
On Dec 13, 2007, at 5:53 PM, Diego Andrés Asenjo González wrote
Hi All;
For me, I am in Kuwait and using the TDM22B and I used
all the below settings and did not resolve my problem,
I do not know if there is any other settings, or if
there is a method to detect that no signaling is still
existed on the, so we can do Hanup, the below settings
used and did not r
Hi Abel,
> Is there a way to catch de gtalkID of a caller that´s calling my
> asterisk gtalk account?
the caller id is not properly set, only its ANI part is. I just
proposed a patch in order to retrieve the CALLERID(name) variable from
the Dialplan, see http://bugs.digium.com/view.php?id=11549.
On Fri, 14 Dec 2007 06:01:49 +0100, Vincent wrote:
>On Fri, 14 Dec 2007 14:50:28 +1000, [EMAIL PROTECTED] wrote:
>>Erm, there just might be, take a look at this...:
>
>Ah yeah, forgot about $angoma ;-) I'll restate this as: No card for
>home/SOHO use, ie. in the $50-100 range for the single FXO po
Hello and thank you for reply... I tried with Playback() and is the same
effect. Is curious because sometime there's no pause other time is a long
pause.
Anybody have other idea?
Thank you.
On 12/14/07, Atis Lezdins <[EMAIL PROTECTED]> wrote:
>
> On 12/14/07, Catalin S. <[EMAIL PROTECTED]> wrote
On Friday 14 December 2007 00:41:18 John Fawcett wrote:
> I've have installed a new Asterisk 1.4.15 system after having previously
> used a 1.2 CVS head (from 10 Sep 2005). Both systems are pentiums though
> the newer one is actually a slower processor.
>
> On the new system, playback of gsm files
I will be out of the office from Dec. 10-13.
Thanks,
Dan Baker
Ursuline Academy
314-984-2828
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John Fawcett wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> I've have installed a new Asterisk 1.4.15 system after having previously
> used a 1.2 CVS head (from 10 Sep 2005). Both systems are pentiums though
> the newer one is actually a slower processor.
>
> On the new system, playba
Dear all
I have asterisk every time my Agent login in queue and useing
queue but i want to staticly map that agent in queue so how do it possible and
what configuration required for it ???
PGP Signature--
Satish Patel
mobile:- +91-9818875535
http://www.linuxbug.org
-
On 12/14/07, Catalin S. <[EMAIL PROTECTED]> wrote:
> Hello,
> i have a simple but annoying problem. I have the following entry in
> /etc/asterisk/externsions.conf file:
>
> --
> exten => 10100,1,Wait(4)
> exten => 10100,2,Playback(transfer,noanswer)
> exten => 10100,3,Dial(${PHONE30},30,t)
Hello,
i have a simple but annoying problem. I have the following entry in
/etc/asterisk/externsions.conf file:
--
exten => 10100,1,Wait(4)
exten => 10100,2,Playback(transfer,noanswer)
exten => 10100,3,Dial(${PHONE30},30,t)
exten => 10100,4,Background(extension)
exten => 10100,5,Background(is-
Dears,
Here is the logs when I put loglevel=255 on unicall.conf, I have use
max-wait = 1
[Dec 14 09:53:42] DEBUG[28143] chan_unicall.c: unicall_call called - 'g1'
[Dec 14 09:53:42] DEBUG[28143] chan_unicall.c: unicall_call caller id -
'3007'
[Dec 14 09:53:42] WARNING[28143] chan_unicall.c: M
Friday, December 14, 2007, 5:47:38 AM, Paul wrote:
> Umm - you could just buy a SPA-3000/3102/3666/etc.
What is SPA-3666?
--
Best regards,
Gergomailto:[EMAIL PROTECTED]
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On Fri, Dec 14, 2007 at 02:55:39AM +0100, Vincent wrote:
> Hello
>
> I was wondering why there doesn't seem to a Windows version of Zaptel,
> making the Digium and its clones unavailable for a Windows PBX.
>
> Is the Zaptel/Zapata combo too *nix-centric?
No. The current zaptel is Linux-centric.
Hello List
I am very interested in developing a research project on security protocol
for VoIP, under the GPL.
For some time I have been reviewing ZRTP, I would like to know the opinion
having regard to whether and under asterisk, but I see that this closed
implementations according am
Http://bu
Hello List.
I just got my new PS3 yesterday, and first thing I did was of course to install
Linux, and then compile asterisk, and it worked without any problems.
My question is this...
Is anybody looking into using the Cell processor for G729 enc/dec?
Using the 6 SPE processing uni
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