On Thu, Jul 3, 2008 at 5:07 PM, Dave Cotton [EMAIL PROTECTED] wrote:
Simon wrote:
Hi There,
Has anyone managed to get 2 AVM ISDN Fritzcard's working in with a 2.6
kernel system?
Yes, with Suse 10.2/10.3 and chan_misdn.
OK. ive got debian etch working with one card compiling the drivers
The Asterisk appliance can currently only auto-provision Polycom phones,
so you're going to need to manually configure the Grandstream phones.
Sydney Web Hosting wrote:
I have 1 nic card which is linked to the router.
Then I use 1 port on the router which is linked to the asterisk appliance.
Hello,
this is the case. Idle goes to 0% and IRQ goes to 100%.
I have a Junghanns ISDN card (bristuff) card. And I guess it is using
that Echo Canceler.
Best regards,
Loic Didelot.
On Thu, 2008-07-03 at 14:52 +1200, Matt Riddell wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Loic
Yesturday I found a bug in Asterisk, in particular in Dial application.
When the Dial function exit it want to branch to n+1, but if n+1 do not
exist, it exit from the context.
Example :
exten = s,5,ChanIsAvail(SIP/604,s)
exten = s,6,Dial(SIP/604,15,wotr)
exten = s,106,NoOp(Matthieu)
exten =
In article [EMAIL PROTECTED],
Kelvin Chan [EMAIL PROTECTED] wrote:
Hi guys,
I'm trying out ooh323 and couldn't bridge ooh323 and sip/zap.
I'm using netmeeting and set gateway to my asterisk.
Here's my CLI dump:
== Spawn extension (h323, , 1) exited non-zero on
If I define a queue like in:
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue
but instead I define queue_table as a MySQL ndbcluster table shared between two
asterisk servers (both active and receiving calls), will the queue calls be
handled coherently (or a pgcluster table if you
Hi
I m making a call from one asterisk server to an asterisk client
The call gets completed but I want it to send dtmf signals
The dialplan I have made for this is like
exten = 205,1,Answer
exten = 205,n,Wait(15)
exten = 205,n,Playback(dtmf-1)
exten = 205,n,Wait(20)
but it does not send any
Hi
Slightly off-topic,
The OLPC (One Laptop Per Child, 100$ Laptop) project has announced a
collection of 10GB of sound samples:
http://wiki.laptop.org/go/Sound_samples
License: CC-BY (explicitly allows public performance for commercial
purpose).
--
Tzafrir Cohen
Use SendDTMF.
--- On Thu, 7/3/08, Neha Punia [EMAIL PROTECTED] wrote:
From: Neha Punia [EMAIL PROTECTED]
Subject: [asterisk-users] (no subject)
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Date: Thursday, July 3, 2008, 10:29 AM
Hi
I m making a call from one
My suggestion is you implement sbs2003 the correct way with a 2 nic
solution.
Yes a 1 nic installation is possible but you lose all the benefits.
Tell whoever set it up to tear it down and implement it properly.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
But if I m using this SendDTMF it does not send anything
I m using it like this in extension.conf
exten = 205,1,Answer
exten = 205,n,Wait(20)
exten = 205,n,Playback(dtmf-1)
exten = 205,n,Wait(20)
exten = 205,n,SendDTMF(9)
exten = 205,n,Wait(5)
exten = 205,n,Read(digito)
Hello everybody
I have configures asterisk server
and i
am using TE220P digium card. Here is the content of
the
/etc/zaptel.conf file
###
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
span=2,2,0,ccs,hdb3
bchan=32-46,48-62
dchan=47
loadzone = in
defaultzone
Hello everybody
I have configures asterisk server
and i
am using TE220P digium card. Here is the content of
the
/etc/zaptel.conf file
###
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
span=2,2,0,ccs,hdb3
bchan=32-46,48-62
dchan=47
loadzone = in
defaultzone
Hi
On Thu, Jul 03, 2008 at 06:21:27PM +0530, Bikrish Amatya wrote:
Hello everybody
I have configures asterisk server
and i
am using TE220P digium card. Here is the content of
the
/etc/zaptel.conf file
###
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
Your E1 links are down. (red alarm) Your card does not like or see your
providers E1.
Lyle
Bikrish Amatya wrote:
Hello everybody
I have configures asterisk server
and i
am using TE220P digium card. Here is the content of
the
/etc/zaptel.conf file
###
Hi all, One question
I have set in the extensions.conf of my asterisk that all incoming call
go in the wait application because I need to not connect the caller
but remain in the ringing state.
After that the call is on the wait exten for a N second I need from
other sip phone to pickup this
chan_iax2 does not support pickup (callpickup=, pickupgroup= and *8).
Enrico Pasqualotto wrote:
Hi all, One question
I have set in the extensions.conf of my asterisk that all incoming call
go in the wait application because I need to not connect the caller
but remain in the ringing
This appears to be a SIP gateway to four FXO ports for ~$250. Has
anybody used it with Asterisk? Comments?
http://www.ipphoneshack.com/products/D_Link_DVG_3104MS_VoiceCenter_4_Port_PSTN_Gateway-193-12.html
Any good reason to pay for a Mediatrix 1204 or some other box instead?
smime.p7s
Hello all,
i recently finished setting up my asterisk with sipura 3102 using PSTN.
this is my dial plan relevant to wht i want:
exten =_01,1,Dial(SIP/$(EXTEN)@200)
right now as u see i made my dial plan on a 2 stage dialing mode.
tht means i dial 01, i get the pstn dial tone, and then i call
On Thu, 2008-07-03 at 09:31 -0500, Eric ManxPower Wieling wrote:
chan_iax2 does not support pickup (callpickup=, pickupgroup= and *8).
Wow! It's a very nice problem
And for redirect a call in wait state to a sip phone? Without pickup ...
Channelredirect don't work with ringing channel for
Tariq,
I cannot see the context= line in your sip.conf setup. Do you have the
appropriate context defined in your sip.conf that match your users context
in extension.conf???
On Tue, Jul 1, 2008 at 7:09 PM, Tariq .. [EMAIL PROTECTED] wrote:
Greetings..
i have 20 extensions with two queues.. i
Hey!
i'm facing the same prob..
i bought an HTC vox (s710) 2 weeks ago, and im still looking for a sip
client..!
so far i found these 3:
AGEphone mobile: http://www.ageet.com/
SJphone: http://www.sjlabs.com/sjp.html
Bria Mobile: http://www.counterpath.com/enterprise-mobility-gateway.html
The number one skill for setting up asterisk is learn how to
communicate since it's a communication application :P
As for your problem looks like you are trying to use the wrong span
for dial out.
On Thu, Jul 3, 2008 at 8:50 AM, Bikrish Amatya [EMAIL PROTECTED] wrote:
Hello everybody
I
Hi Roland,
Did you try:
http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
ws-mobile-6x-for-free-voip-calls-using-asterisk/
We have this successfully working on a Touch (ELF), and a HTC Tilt (Tytn II)
Thanks,
Matt G
: http://www.voipphreak.ca
:
On Thu, 3 Jul 2008, Alex Balashov wrote:
C F wrote:
The number one skill for setting up asterisk is learn how to
communicate since it's a communication application :P
Oh, if only more newbie posters on this list would heed that advice.
) How about rejecting emails that don't have a
I have two different scenario where I would like to apply different tones:
1. Incoming calls should have a different ringing tone than transfer
calls
2. While on a call, transfer calls should have a different beep sound
on the handset than incoming calls.
How/where can I
Hey Matt!!
thanks for the advice!
appreciate it.. just installed it and everything worked fine ( i got internet
calling in my menu) though i cant seem to access the editing tool..
keeps on giving me some error even after soft reseting..
any idea?!
From: [EMAIL PROTECTED]
To:
On Thu, 3 Jul 2008, Alex Balashov wrote:
Steve Edwards wrote:
On Thu, 3 Jul 2008, Alex Balashov wrote:
C F wrote:
The number one skill for setting up asterisk is learn how to
communicate since it's a communication application :P
Oh, if only more newbie posters on this list would heed that
Steve Edwards wrote:
On Thu, 3 Jul 2008, Alex Balashov wrote:
Steve Edwards wrote:
On Thu, 3 Jul 2008, Alex Balashov wrote:
C F wrote:
The number one skill for setting up asterisk is learn how to
communicate since it's a communication application :P
Oh, if only more newbie posters on
Alex Balashov wrote:
Steve Edwards wrote:
On Thu, 3 Jul 2008, Alex Balashov wrote:
Steve Edwards wrote:
On Thu, 3 Jul 2008, Alex Balashov wrote:
C F wrote:
The number one skill for setting up asterisk is learn how to
communicate since it's a
So who out there is aware of the FCC or FTC laws concerning spoofing caller
ID for deceptive purposes? There's a collection agency out there who has my
wife's name crossed with someone else's, and they are picking numbers from
our area code to present themselves as when calling us (over and over
any body know about a iax softphone for palm os ?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To
On Thu, Jul 3, 2008 at 9:26 PM, Robert Goodyear [EMAIL PROTECTED] wrote:
So who out there is aware of the FCC or FTC laws concerning spoofing caller
ID for deceptive purposes? There's a collection agency out there who has my
wife's name crossed with someone else's, and they are picking numbers
Yeah I'm thinking either homeland security or some other identity-critical
legislation might be on my side here.
On Thu, Jul 3, 2008 at 12:40 PM, randulo [EMAIL PROTECTED] wrote:
On Thu, Jul 3, 2008 at 9:26 PM, Robert Goodyear [EMAIL PROTECTED]
wrote:
So who out there is aware of the FCC or
On Fri, 4 Jul 2008, Peter Lindquist wrote:
Steve Edwards wrote:
But deciphering posts from our non-English-speaking members is half the
challenge/fun :)
Seriously though, good for them for trying. I wouldn't.
What are you if you speak 3 languages? Trilingual.
What are you if you
Robert Goodyear wrote:
Yeah I'm thinking either homeland security or some other
identity-critical legislation might be on my side here.
On Thu, Jul 3, 2008 at 12:40 PM, randulo [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
On Thu, Jul 3, 2008 at 9:26 PM, Robert Goodyear
[EMAIL
Neither DHS nor FTC has any legislation on this. Florida house had a
bill. Unfortunately, Collection agencies are deceptive by nature as most
other options have been exhausted before an account goes to collections.
I get the same thing here; they once even called me from a number that
had my same
So, I'm trying to get the Asterisk vxml (from i6net) working.
Having no luck with it.
My dial plan has:
exten = _X.,1,Answer()
exten = _X.,n,Wait(1)
exten = _X.,n,Vxml(file:///tmp/menu.vxml)
The /tmp/menu.vxml file has:
?xml version=1.0?
vxml version=1.0
form
blockaudio
Hi Roland,
No problem, glad this works for you. We don't find it too bad.
Hm, I'm not sure why you're having difficulty with the editing tool, you can
check on xda-developers.org forum for more information on the editing tool,
there may be a newer version. If you need help, feel free to
Eric ManxPower Wieling wrote:
Make the card stop sharing it's IRQ with your IDE controller. Try
moving the card to another slot.
Asterisk has to send an audio packet every 20ms for VoIP calls. I
believe Zaptel expects no more than a few ms of latency. If something
is causing a delay,
Does vxml let you use absolute paths?
Wouldn't it have the equivalent of a DocRoot???
Alex
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, July 03, 2008 5:03 PM
To: asterisk-users@lists.digium.com
Not for file:// access, No...
- Original Message
From: Alexander Lopez [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, July 3, 2008 2:21:42 PM
Subject: Re: [asterisk-users] Asterisk VXML... Help.
Does vxml
Alex Balashov wrote:
) How about rejecting emails that don't have a subject?
That is an excellent idea.
If a person doesn't have enough clue to use a subject, then we're really
just feeding the beast when we indulge the question with an answer.
And the archived version of that
On Thursday 03 July 2008 15:54:36 Alexander Lopez wrote:
Neither DHS nor FTC has any legislation on this. Florida house had a
bill. Unfortunately, Collection agencies are deceptive by nature as most
other options have been exhausted before an account goes to collections.
I get the same thing
Snip
On Thursday 03 July 2008 15:54:36 Alexander Lopez wrote:
Snip
There is one Alex Lopez (NOT ME) here in Miami that owes a lot
of
people a lot of money. I get calls at all times of the day and
night,
they forge the number, and so what do they care about following the
FTC
Robert Goodyear wrote:
So who out there is aware of the FCC or FTC laws concerning spoofing
caller ID for deceptive purposes? There's a collection agency out
there who has my wife's name crossed with someone else's, and they are
picking numbers from our area code to present themselves as
Alexander Lopez wrote:
I may create an IVR Hell for them, so that I can transfer the calls to,
Hey, Its their dime.
If you do end up going that route, please share the details, and
possibly the code. We could all benefit from a good IVR from hell. I
certainly could use one.
No, I'm not
On Fri, Feb 22, 2008 at 10:15 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Friday 22 February 2008 04:55:13 Vincent wrote:
On Thu, 21 Feb 2008 22:04:41 +0200, Tzafrir Cohen wrote:
For the brave: use modules.conf without 'autoload = yes'. This promises
you many hours of interesting dialplan
Hello All,
We need the DIDs of Paris and Gottenburg, Sweden. Can anyone provide ?
Please reply with rates.
Regards,
--
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com
Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766
Email: [EMAIL PROTECTED]
MSN: [EMAIL
Hi Kashif,
I use didx.net you can get did numbers in many countries
On Jul 3, 2008, at 10:25 PM, Kashif Naeem wrote:
Hello All,
We need the DIDs of Paris and Gottenburg, Sweden. Can anyone
provide ? Please reply with rates.
Regards,
--
Kashif Naeem
Business Development Manager
Hadi
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