Steve Edwards wrote:
> On Tue, 4 Dec 2007, Philip Prindeville wrote:
>
>
>> I wanted to write a "popcorn" app for myself, both to learn how to script in
>>
>
> Just out of curiosity, what does this have to do with popcorn?
>
> Thanks in advance,
> ---
I'm sure this has been asked a million times before, but is there an easy
wa to have Asterisk register more than one (distinct) contact binding
concurrently?
The goal is to have two phones register with the same credentials from
different locations and consistently and reliably ring on inbound
On Tue, 4 Dec 2007, Philip Prindeville wrote:
> I wanted to write a "popcorn" app for myself, both to learn how to script in
Just out of curiosity, what does this have to do with popcorn?
Thanks in advance,
Steve Edwards
Hi.
I wanted to write a "popcorn" app for myself, both to learn how to
script in extensions.conf, but also because it was something handy.
Along the way, I found myself doing something like:
[popcorn]
exten => s,1,Set(FUTURETIME=$[${EPOCH} + 10])
...
exten => s,n,While(${EPOCH} < ${FUTURETIME
Yes - tomorrow night is the monthly Asterisk meeting held by the
Melbourne Asterisk group. (Melbourne, Australia that is)
Venue is usually Pint on Punt (corner of Punt Rd and Peel Street) from
7pm onwards.
Feel free to turn up, eat food, drink beverages and talk about Asterisk.
later,
Is this getting through??
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Everton,
This sounds like the app_rxfax module has a dependency on some other
module which implements T.30 handling, and that this module is either
not loaded, or that its symbol table is not being shared in the monolithic
core.
-- Alex
On Tue, 4 Dec 2007, Everton Goularth wrote:
Hi people
I have the extension connected to the fxs on the x400p (2 modules) and
I use *0 which is actually built into the code to flash the fxo line.
Hope this helps.
on Tuesday 12/04/2007 C F([EMAIL PROTECTED]) wrote
> application map in features.conf
>
> On 12/4/07, Patricio Valarezo Lozano <[EMAIL
Replying to myself
Its fixed now
Checking timestamps is optional according to RFC so asterisk is not
doing it.
Anyway, I made a patch and tested it and its working.
Thanks.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Boris
application map in features.conf
On 12/4/07, Patricio Valarezo Lozano <[EMAIL PROTECTED]> wrote:
> Hi, I hope someone could help me, i have a x100p interface for testing
> purpose and on each incomming call I redirect the call to handytone 388
> atas, the problem comes when i'm during a call and
Hi Alan,
Thanks for these helpful comments.
I had a look at David Rowe's site, and found it very interesting - I'm
blind myself, and the Louder router project he's working on sounds like a
wonderful idea.
It looks like I got my wires crossed, to some extent - I hadn't realised
that all the ec
Matthew Fredrickson wrote:
> This looks like a really good reason to call Digium tech support :-)
> It's comes free with the purchase of the card. I haven't heard of
> anything like this, although posting your kernel panic output would
> help. But it would be best to handle this through tech s
Hi, I hope someone could help me, i have a x100p interface for testing
purpose and on each incomming call I redirect the call to handytone 388
atas, the problem comes when i'm during a call and another call comes
in, i hear the call waiting beep (comming from the zap channel), but I
can't catc
Hello,
Which server do you choose for holding a couple of TDM2400 (2 of them) ?
I would like to install it in a not too deep rack (< 60 cm).
Regards
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Edwin Lam wrote:
> hi folks.
>
> i have a Digium TE220 PCI-E 2 port T1/E1 controller installed
> in an IBM x3400 server. i load the wct4xxp driver seems ok.
> but when i execute "ztcfg -vvv" command. the kernel panic.
> i tried zaptel 1.2.21 & 22. they have the same result.
> following is my zapte
Hi people,
I'm tring to configure fax on my asterisk server. I'm using the
instructions of: http://www.asteriskguru.com/tutorials/spandsp.html and
the files app_rxfax.c, app_txfax.c and apps_Makefile_patch from:
http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/
I h
Greetings, List.
I would like to implement a procedure in my call center but am not sure
the best way to implement it. I'm hoping I can describe it here and
that I'll receive some feedback and/or suggestions on how to proceed.
Here's my situation:
My call center fields calls regarding interne
Does someone know why the posts from some users on Usenet are just one
long line, with no carriage return?
On Tue, 4 Dec 2007 17:02:12 +, dadsadsadf dsadasdsa
<[EMAIL PROTECTED]> wrote:
>Hi all, I want to use Asterisk as an IVR system.
O'Reilly's "Asterisk, the future of telephony" doesn't ha
Nikhil Nair wrote:
> Hi,
>
> I'd like to try using a good quality microphone and a set of PC speakers
> (in the first instance) to create a powerful speakerphone; if I get that
> working, I'll probably try more elaborate audio equipment.
>
Interesting... After playing with - and being very imp
In my other response to this topic I mentioned chan_mobile, I could have
meant chan_bluetooth.
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Nikhil Nair wrote:
> In addition, I've tried using 'dial' from the Asterisk console while a
> call from the console is already established, hoping that this would send
> DTMF signals. So far those signals haven't been received at the other
> end, while I've had no trouble sending DTMF from a Wi
Nikhil Nair wrote:
> I gather Asterisk can do very good software echo cancellation, but I can
> see no reference to using it with chan_console (using the Alsa driver).
> Am I overlooking something obvious, or is that really not implemented?
>
IIRC, echo cancellation is fully in zaptel. If you
Jared Smith wrote:
> On Tue, 2007-12-04 at 08:24 -0800, ticket john wrote:
>
>> Please active my account public, thanks admin.
>>
>
> I'm not sure I understand what account you're talking about. If you're
> talking about your subscription to the mailing list, you're already
> subscribed.
Hi,
I'd like to try using a good quality microphone and a set of PC speakers
(in the first instance) to create a powerful speakerphone; if I get that
working, I'll probably try more elaborate audio equipment.
For this to work, I'll need software acoustic echo cancellation, or the
caller at the
It appears that this has been fixed on the SPA-942 but not the 941.
Hopefully that will come soon. Thanks.
Doug
> [EMAIL PROTECTED] On Behalf Of Paul Hales
> Sent: Tuesday, December 04, 2007 12:46 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Ph
Thanks for this explanation : it's now crystal clear !
2007/12/4, Steve Edwards <[EMAIL PROTECTED]>:
>
> On Tue, 4 Dec 2007, Olivier wrote:
>
> > Can anyone explain the difference between Asterisk Gateway Interface and
> > Asterisk Management Interface ? Is it correct to consider AGI scope to
> >
Hi all, I want to use Asterisk as an IVR system. I have read in the web some
dialplan examples for menus, like the following example: [default]exten =>
steve,1,Dial(SIP/steve);exten => mark,2,Dial(SIP/mark);[mainmenu]exten =>
s,1,Answerexten => s,n,Background(thanks) ; "Thanks for calling press
On Tue, 2007-12-04 at 08:24 -0800, ticket john wrote:
> Please active my account public, thanks admin.
I'm not sure I understand what account you're talking about. If you're
talking about your subscription to the mailing list, you're already
subscribed.
-Jared Smith
On Tue, 2007-12-04 at 17:20 +0100, Marc LEURENT wrote:
> It's just that I received SIP notify message saying that there is
> nothing in the voicemail even when there is a message...
Do you have a mailbox defined for the SIP device in sip.conf? If you
don't, Asterisk has no way of matching up a ma
Please active my account public, thanks admin.
Best Regards,
-
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
It's just that I received SIP notify message saying that there is
nothing in the voicemail even when there is a message...
my voicemail.conf
[default]
; Define maximum number of messages per folder for a particular context.
;maxmsg=50
; Mailboxes ma
On Tue, 4 Dec 2007, Lees, James (UK) wrote:
>
> Hello!!!
>
> I am using asterisk with a usb headset and therefor no actual telephony
> handset.
>
> I have configured asterisk to use voicemail and I can dial the mailbox
> via my touchscreen application. The only problem is voicemail asks you
> to p
You can disable call waiting in the eyeBeam.
On Tue, 4 Dec 2007, Joao Pereira wrote:
Hello
I'm using eyeBeam, and Asterisk keeps sending my clients a second call,
when they are still in one call (because eyeBeam has lots of channels).
I was using X-Lite (with 3 channels) and Asterisk never sen
Hi,
>However, I believe that zaptel >= 1.4.6 or zaptel 1.2 >= 1.2.21 should
>support hires timers for timing on kernel >= 2.6.22 .
>
>What version of Zaptel do you use?
>
I was using version 1.4.5.1
I just downloaded and installed version 1.4.7, configure/make/make
install finished without
Hello!!!
I am using asterisk with a usb headset and therefor no actual telephony
handset.
I have configured asterisk to use voicemail and I can dial the mailbox
via my touchscreen application. The only problem is voicemail asks you
to press "1" to hear messages. With a handset this is simple, bu
We are using Ethernet Bonding with no problems at all. Each server has 2
build-in NIC's and a quad NIC. They are divided into 3 networks with 2
NIC's in each. Links are up on all 6 connections and you don't even hear
a click if I unplug the 'live' ethernet. 3 different networks on 3 or
more NIC
Pepo wrote:
> Hi friends.
>
> I have problems with the voicemail system, when some user "forward" the
> message to other box all the Asterisk falls down and restart.
>
> How do I disable the option to forward messages in voicemail (option 8 in the
> menu)? and Which can be the cause for the pro
Hi,
I have a SIP provider who sometimes sends duplicate RTP packets to me.
Sent RTP packet to 10.55.20.201:17440 (type 08, seq 008536, ts
4846560, len 000160)
Got RTP packet from10.55.20.201:17440 (type 08, seq 051978, ts
3647104992, len 000160)
Got RTP packet from10.55.20.
Tzafrir Cohen wrote:
> The proper way to set module parameters is in /etc/modprobe.conf or
> in a file under /etc/modprobe.d (depending what your distribution uses .
> modprobe of 2.6 can use either). Put there the line:
>
> options wct4xxp t1e1override=15
Like so:
echo 'options wct4xxp t1e1o
Hello All
What are the best system specifications for 60 SIP extensions. Also about 15
of them can be in conference at a time.
Regards,
--
Kashif Naeem
Director
Hadi Telecom
www.haditelecom.com
Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766
Email: [EMAIL PROTECTED]
MSN: [EMAIL PROTECTED]
Hello All
How is Digium Support for FXO Cards or any other devices ? Can someone
compare it with Sangoma ?
Regards,
--
Kashif Naeem
Director
Hadi Telecom
www.haditelecom.com
Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766
Email: [EMAIL PROTECTED]
MSN: [EMAIL PROTECTED]
Gmail: [EMAIL PROTE
Tzafrir Cohen wrote:
> On Mon, Dec 03, 2007 at 09:56:51PM +0100, Philipp Kempgen wrote:
>> Tilghman Lesher wrote:
>>> modprobe wct4xxp t1e1override=15
>>>
>>> t1e1override is a bitwise parameter, 0 being all T1, 15 being all E1 and
>>> numbers in between as different combinations of T1 and E1 for
On Tue, 4 Dec 2007, Stefan Guenther wrote:
> For whatever reason I have to insert a WAIT(1) in front of every
> application that returns an output. Well, now the context looks like
> this and it works:
>
>exten => 202,1,ANSWER()
>exten => 202,2,WAIT(1)
>exten => 202,3,PLA
On Tue, 4 Dec 2007, Olivier wrote:
> Can anyone explain the difference between Asterisk Gateway Interface and
> Asterisk Management Interface ? Is it correct to consider AGI scope to
> focus on call handling and AMI scope to anything which can be done with
> Asterisk froma loading new modules t
>> Hello
>> I'm using eyeBeam, and Asterisk keeps sending my clients a second call,
>> when they are still in one call (because eyeBeam has lots of channels).
>> I was using X-Lite (with 3 channels) and Asterisk never sent the client
>> a second call.
>>
>> How can I force Asterisk (or eyeBeam) jus
Joao Pereira wrote:
> Hello
> I'm using eyeBeam, and Asterisk keeps sending my clients a second call,
> when they are still in one call (because eyeBeam has lots of channels).
> I was using X-Lite (with 3 channels) and Asterisk never sent the client
> a second call.
>
> How can I force Asterisk (
Hi,
I have found a solution for my problem or at least I can hear the output
of PLAYBACK() and the voicemail system.
Since some of you suggested a timing problem, I removed the ztdummy and
zaptel modules, but this had no effect.
For whatever reason I have to insert a WAIT(1) in front of every
ap
On 12/4/07, Joao Pereira <[EMAIL PROTECTED]> wrote:
> Hello
> I'm using eyeBeam, and Asterisk keeps sending my clients a second call,
> when they are still in one call (because eyeBeam has lots of channels).
> I was using X-Lite (with 3 channels) and Asterisk never sent the client
> a second call.
Hello
I'm using eyeBeam, and Asterisk keeps sending my clients a second call,
when they are still in one call (because eyeBeam has lots of channels).
I was using X-Lite (with 3 channels) and Asterisk never sent the client
a second call.
How can I force Asterisk (or eyeBeam) just to send one call
> Jsut a dumb question: are all four ports connected to the telco? If not:
> which of them is?
>
Yes the first 3 spans are connected to de teleco, the last one no.
> >
> > Now I'm talking with Technical support from teleco operator (Brasil
> > Telecom) to confirm de cable standard.
> >
Great !
Yes, the log file..
El Martes, 4 de Diciembre de 2007 12:01, Newbie escribió:
> Hello,
> could you please advise .. where can I find the trace of asterisk? do you
> mean log file?
>
> Thanks & Regards
> Bie
>
>
> - Original Message -
> From: "Guillermo Rodriguez" <[EMAIL PROTECTED]>
> To:
Hello,
could you please advise .. where can I find the trace of asterisk? do you
mean log file?
Thanks & Regards
Bie
- Original Message -
From: "Guillermo Rodriguez" <[EMAIL PROTECTED]>
To: "Newbie" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commercial Discussion"
Sent: Tue
Can you put the trace of asterisk.??'
When you call to 988
Thx.
Guillermo
El Viernes, 30 de Noviembre de 2007 10:17, Newbie escribió:
> Dear The Expert,
>
> I am very new with this, I have installed AsteriskNow, X-Lite as my
> SoftPhone, I am using SPA-3102.
> I have 3 extensions,
>
> me at 2
On Tue, Dec 04, 2007 at 07:06:14AM -0200, Roger C. Beraldi Martins wrote:
> 2007/12/4, Tzafrir Cohen <[EMAIL PROTECTED]>:
> > On Mon, Dec 03, 2007 at 10:51:43PM +0100, Philipp Kempgen wrote:
> > > Richard Lyman wrote:
> > >
> > > > I have never noticed, does the output of ztcfg change is it set to
Hi
Is there a way to catch de gtalkID of a caller that´s calling my
asterisk gtalk account?
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Hi,
Can anyone explain the difference between Asterisk Gateway Interface and
Asterisk Management Interface ?
Is it correct to consider AGI scope to focus on call handling and AMI scope
to anything which can be done with Asterisk froma loading new modules to
originating calls ?
Regards
___
2007/12/4, Tzafrir Cohen <[EMAIL PROTECTED]>:
> On Mon, Dec 03, 2007 at 10:51:43PM +0100, Philipp Kempgen wrote:
> > Richard Lyman wrote:
> >
> > > I have never noticed, does the output of ztcfg change is it set to E1?
> >
> > Yes. More channels. :)
>
> No. The channels listed in ztcfg -vv are the
The "nat=no" did help fix the problemThanks!
John
> Date: Mon, 3 Dec 2007 18:04:29 -0800> From: [EMAIL PROTECTED]> To:
> asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Problem
> registering Cisco 7970 phone with Asterisk 1.4 running FreePBX> > John
> Constalgie wrote:> > > > H
On Mon, Dec 03, 2007 at 10:59:45PM +0100, Stefan Guenther wrote:
> Hi,
>
> >My quick guess would be that it's a timing issue. You didn't mention
> >whether you are using a Zaptel device or ztdummy.
> >
> I'm using ztdummy, and yes, I guess your're right - it seems to be a
> timing problem, be
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