On Sun, 2005-09-25 at 08:58 +0300, Tzafrir Cohen wrote:
On Sat, Sep 24, 2005 at 10:08:19PM -0700, trixter http://www.0xdecafbad.com
wrote:
Has anyone built a game with the dialplan? I would think this would
most easily be managed by an AGI, but its possible with realtime
extensions.
On 9/24/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
My Asterisk book is on its way, so please bear with me.
My experience so far is limited to sip.conf and extensions.conf, as I
don't have a hardware board yet.
First: It seems like an extension can be part of more than one context?
An
On Sat, 2005-09-24 at 23:30 -0700, trixter http://www.0xdecafbad.com
wrote:
On Sun, 2005-09-25 at 08:58 +0300, Tzafrir Cohen wrote:
On Sat, Sep 24, 2005 at 10:08:19PM -0700, trixter http://www.0xdecafbad.com
wrote:
Has anyone built a game with the dialplan? I would think this would
On 23:46, Sat 24 Sep 05, Sascha Ferley wrote:
Hi,
I was wondering if it is possible to setup with Asterisk a Cisco 7960 to use
extension mobility / roaming.
Meaning that a user logs into a phone and his profile moves with him / her.
I have a network of ~75 Cisco 7960 phones, running
Hi! I asked this question a couple of days ago but
got no answer so I try again.
Is it possible to route a call in * based on used
codec, meaning that if a user use G723 that call is routed to siptrunk 1 and a
user using G.729 is routed to siptrunk 2?
Regards
Anders Svensson
Chris Coulthurst wrote:
So the IP 601 is the 600 with a few extras? Looks like Polycom
dropped the ball again -- yet another pretty phone with NO BACK
LIGHT. Does the design team at Polycom have their brains unscrewed?
When I was at Spring VON 04, I was talking with the guy from Polycom
Mir ha scritto:
Thanks for your answer.
This is not what the customer wants, they answer +500 calls a day, and
dont want to say Welcome to BigCorp every time.
They want a personal welcome file to be played to the caller every
time they pick up the ringing phone.
Maybe you can do a quick
Just for your Information , Digium Card FXO and FXS four port each available
in Dhaka. Bangladesh, Try to contact with Mr. Delwar.
Thanks.
Bashir
- Original Message -
From: Capt MS [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi:
My experience with Areski is I wasn't able to get it
to work and wasn't able to get help including from the
owner of idiot guide who inturns wasn't able to get
areski to work either according to him.
I easily downloaded astcc and works fine
Regards;
Chawki Hammoud
--- ADEGOKE ARUNA
Termilink Digital Voice www.termilink.net
On 8/13/05, jonny hashem [EMAIL PROTECTED] wrote:
what is the best voip provider that provides goodservice ,good voice quality and good rates . any one
havean experience with voip providers advice
There is also www.talkycallshops.com
Very good rates, no
monthly fee Unlimited number of numbers. Voip DIDs in 22+ countries.
Anders
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nana Tandoh
Sent: den 25 september 2005 12:20
To: Asterisk Users Mailing List
On Sun, 2005-09-25 at 06:20 -0400, Nana Tandoh wrote:
Termilink Digital Voice www.termilink.net
On 8/13/05, jonny hashem [EMAIL PROTECTED] wrote:
what is the best voip provider that provides good
service ,good voice quality and good rates . any one
have an
AreskiCC works great for me , i've been using it for ~ 500 + cards scene and it works awesome for me! really , the guy did a REALLY good job , trust me.
cheers
~uppal
On 9/25/05, chawki hammoud [EMAIL PROTECTED] wrote:
Hi:My experience with Areski is I wasn't able to get itto work and wasn't
Hi
I've 3 iax connections to my provider , each of them have own DID ,
PH1|
|
\/
PH2--|-| --- ||-- DID1
| A1 | --- |ISP |-- DID2
PH3--|-| --- ||-- DID3
I had iax phone on each of
Thanks maka and sorry I just saw your email (too many in my account). I set
relaxdtmf=yes and going to try it. Thanks again.
Larry
--
Message: 5
Date: Mon, 5 Sep 2005 12:48:28 +0300
From: maka [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] DTMF issue on IVR
To:
I've 3 iax connections to my provider , each of them have own DID ,
PH1|
|
\/
PH2--|-| --- ||-- DID1
| A1 | --- |ISP |-- DID2
PH3--|-| --- ||-- DID3
I had iax phone on each
Dear all.Okay, I'm about to give up now as this issue has been driving me insane for a whileI have an asterisk server (which I have just updated to 1.2beta1).Several phones are connected on LAN, no problemPeople on the go using either x-lite or eyebeam without a problemI have a Sipura 3000
I`m trying a skype usb gateway pluged to a asterisk FXO port.
It's working well, but have to leave a winxp machine online.
http://www.skyvoice.com.br/
But, http://www.voip-info.org/wiki-Skype+Gateways looks better,
integrating skype to sip without using a fxo port to integrate
asterisk.
--
Hi
I have registered a number with an external sip proxy which provides a
real telephone number.
I configured * to forward this number to an internal extension, this works.
Now, I am trying to forward the external number for incoming calls to a
macro, which includes Authenticate(). The
What does zttest tell you?
Well on the 2.4 box it tells me (see below output).
I've not really found any documentation on zttest on what is good or
what is bad so at this point the numbers don't mean much.
Also curiously enough (zttest) is not available on the 2.6 kernel system
Both systems are
On 9/25/05, trixter http://www.0xdecafbad.com [EMAIL PROTECTED] wrote:
what is the best voip provider that provides good
service ,good voice quality and good rates . any one
have an experience with voip providers advice me.
How do you define good service? tech
You might be able to do this in CVS head Asterisk with the SIP_HEADER
variables and a agi script.
Need to look in the source code.
-bill
On 25-Sep-05, at 3:48 AM, Anders Svensson wrote:
Hi! I asked this question a couple of days ago but got no answer so
I try again.
Is it possible to
Thanks for replying Alvaro but for me that same dialplan is not working. I
know the digits are sent as the calling card menu doesn't come up and
instead after a few seconds it says the number you have dialed is invalid.
I have tried putting in multiple w's but like I said, the problem is that
the
whichbrandofMTA are you using? have you tried an ATA or IAD like Grandstream or Sipura?
On 9/24/05, Calvin Lockhart [EMAIL PROTECTED] wrote:
Hi gang,I've been trying to use asterisk with an MTA device can any one offer some help as to how asterisk can work with the thing.
thanks a
Hi,
I hope some can help me out, as I have a problem with the vpb driver in *,
and I am hoping someone has hit these issues and can offer some advice.
First off, I'm connected to a British Telecom SystemX exhange with the local
Tel Co. I can ring out fine, but when the callee on POTS
This is probably generic to any device... However..
Incoming callerid is working with number only.
If I try for any reason to use the function Set(Callerid(name)=blah
blah) it will then send only the outgoing extension as the callerid to
the phone that i s connected to the sipura device...
I
Does anyone on the list have a script for notifying pagers that they
would be willing to share? I have found a reference in the archive to
such a script, but previous attempts to find the author of that
posting have failed.
Anyhow, I am looking to set up a system whereby a message is sent
On 9/25/05, Brian Capouch [EMAIL PROTECTED] wrote:
Website at http://www.mixnetworks.net.
Either this is the wrong URL, or their server is down. I get Connection
Refused
Bad timing :-)
Gah! My bad.
Its http://www.mixnetworks.com - not .net.
Sorry!
--
Leif Madsen -
On Sun, Sep 25, 2005 at 11:26:19AM -0400, Steve Gladden wrote:
What does zttest tell you?
Well on the 2.4 box it tells me (see below output).
I've not really found any documentation on zttest on what is good or
what is bad so at this point the numbers don't mean much.
Also curiously enough
On Sun, 2005-09-25 at 14:37 -0400, Leif Madsen wrote:
Its http://www.mixnetworks.com - not .net.
Sorry!
They dont seem to have their rate information easily locatable, and I am
afraid of voip companies that hide their rates. They have a pdf (I get
only a blank page) for the 'local calling
Yes it works, the only thing is that you need to patch you asterisk for support R2
On 9/23/05, Alex Kauffmann [EMAIL PROTECTED] wrote:
We have several in operation but with isdn and not R2. I know I've seen emails from people that use them with Telmex and have them operating, albeit with some
Hi,
I'm trying to set up Asterisk behind my WRT54GP2 router that has a
intergrated ATA box.
My box are not locked in any way so I can access and change all settings.
Now to the problem...
I have gotten Asterisk to register with my provider and everything works
just well..
Now it's time to get
VoicePulse! (www.voicepulse.com)
I've been extremely pleased with the quality of the VOIP calls, uptime,
and on the rare occasions I've needed support, they responded within 1
day and followed through great on the tickets. Rates are exceptional
too, national calling 200 minutes is $15/mo and
On 9/25/05, Anders Svensson [EMAIL PROTECTED] wrote:
There is also www.talkycallshops.com
That looks interesting. Do they offer iax service or sip only? Do
you have any .conf example?
___
--Bandwidth and Colocation sponsored by Easynews.com --
I use Teliax and they support IAX. http://www.teliax.com/?a=64254372
Regards,
Chris
- Original Message -
From: Shawn Rutledge [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, September 25, 2005 4:08 PM
For the moment only sip. But they will set up an * to be able to provide IAX
too but it will take some time yet.
Mail to [EMAIL PROTECTED] and explain what you want and they will get
back to you tomorrow. Its bedtime in Sweden now ;-)
Anders
-Original Message-
From: [EMAIL PROTECTED]
On 9/25/05, trixter http://www.0xdecafbad.com [EMAIL PROTECTED] wrote:
On Sun, 2005-09-25 at 14:37 -0400, Leif Madsen wrote:
Its http://www.mixnetworks.com - not .net.
Sorry!
They dont seem to have their rate information easily locatable, and I am
afraid of voip companies that hide their
what I do is loopback the WAN port to a LAN port and am able to use
both (ie) take a cable from the wan port of the router and plug it into
the lan port on the same router. This will give you a local ip
and it still should allow connection out to your other provider.On 9/25/05, Johannes [EMAIL
Actually, just point the line you want to use to a local ip
address (the asterisk server). I currently do this with my service. i.e. If your
Asterisk server is 192.168.15.200, just make the proxy for line1 that
address. It routes internally just fine.
Sherwood McGowan
From: [EMAIL
I am also looking for this functionality for an emergency support number.
Except, I want to notify a different person after 7 minutes, then a third,
then back to the first, etc. until the message has been listened to.
--
--
Steven
May you have the peace and freedom that come from abandoning
On Sun, Sep 25, 2005 at 03:04:31AM +0200, Stefan de Konink wrote:
On Sun, 25 Sep 2005, Marco Supino wrote:
I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my
problem is that the BIOS assigns the same IRQ to the SCSI controller,
and the TDM400P, i have tried
Which file does the jitterbuffer setting go in, zaptel or zapata.conf?
I can't find it documented anywhere. What version of zaptel drivers include a
jitterbuffer?
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
In zapata.conf
; Configure jitter buffers in zapata (each one is 20ms, default is 4)
;
jitterbuffers=16
Alejandro Ghergherian
-Mensaje original-
De: Rod Bacon [mailto:[EMAIL PROTECTED]
Enviado el: Domingo, 25 de Septiembre de 2005 08:32 p.m.
Para: Asterisk Users Mailing List -
If anyone is interested in some used digium hardware for their projects:
T-1 card:
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5101873998
4 port FXO cards:
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5102115738
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5102264062
What ports does the Cisco 7960 phones require to be opened in my firewall?
Thanks
Keith
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I'm using CVS HEAD, downloaded a couple of days ago (how do you get
the version anyway?). Everything seems to work great except I cannot
transfer calls that I place. Incoming calls are no problem.
I have features.conf to do blind transfers with *1. When a call comes
in, I can press *1 and I
Did you find a solution to this?
Kib Eki wrote:
yes, fedora 3 but without any changes at the sources
Master Abi wrote:
Are you using Redhat/Fedora? If I remember those init scripts is for
Redhat/Fedora. I am using gentoo.
Did you make any modifications to wct4xxp.c. or pass any parameters
Is anyone offering a vonage-like service using a 100% asterisk only
solution? Just for curiosity.
Thanks,
Waldo
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
All of the providers given so far seem to have a
limited simultaneous connections. As a business solution (multiple outgoing
calls at one time) what are you guys using?
-Scott
___
--Bandwidth and Colocation sponsored by Easynews.com --
I've recently upgraded [EMAIL PROTECTED] to CVS HEAD and in addition to losing the
ability to use the MySQL database, I've noticed that my MOH has degraded
significantly. I've tried all sorts of remedies -- removing the x100p
card and loading asterisk without the zaptel drivers and such --
On 9/25/05, Scott Wolfe [EMAIL PROTECTED] wrote:
All of the providers given so far seem to have a limited simultaneous
connections. As a business solution (multiple outgoing calls at one time)
what are you guys using?
I've been using a mixture of voxee, voipjet, mutualphone and nuphone.
Teliax has rates of about .04 per minute to Mexico. I've been using
them for termination (and fax and DID)
for a while and I haven't had any trouble with them.They support IAX and
SIP.
Regards,
Chris
- Original Message -
From: Jaime Lopez [EMAIL PROTECTED]
To: Asterisk
Yes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Waldo Rubinstein
Sent: Sunday, September 25, 2005 9:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Vonage-type service
Is anyone offering a
You can set up the users using the agent channel.
Then have a AgentLogin fuction in your dialplan. That
would let the phone ring where they are.
Extension Called ID could be handled with a call to the
Set(CallerID...) functions that could be done as the person dials out in the
dial plan.
I anyone has any hesitations in upgrading their 405P (or 410P) to V2 of the
firmware, read below;
I installed one today (turnaround time around 2 weeks to Australia, inc. economy
freight in both directions... impressive!) and have noticed immediate,
significant improvements.
Audio levels
Rod Bacon wrote:
Audio levels are better (have set tx and rx gains back to 0.0) and
missed frames have gone (popping, clicking, etc.). Echo on bridged calls
has also gone (I have now been able to disable echo cancellation on
bridged calls, too!).
Bridged calls with 2nd gen firmware result
Is there similar changes for the TE110 card?
This new firmware only works on new hardware I guess..
Jean-Yves
On 26/09/2005, at 2:36 PM, Kevin P. Fleming wrote:
Bridged calls with 2nd gen firmware result in the audio never
leaving the card; that's why you are seeing such an improvement.
http://www.telasip.com/index.html?partner=tommy13vOn 9/25/05, Waldo Rubinstein
[EMAIL PROTECTED] wrote:Is anyone offering a vonage-like service using a 100% asterisk only
solution? Just for curiosity.Thanks,Waldo___--Bandwidth and Colocation sponsored
I am wondering -- I have a tdm400p with two modules and I understand
that there is a REV I of the card --- Ihave e/f. I am getting
somepopping and I definitely need echo cancel when bridged. Now would
a rev I help in these matters and how can I get them to replace mine
without or with minimal
Hi,
We don't want a digital receptionist if we can help it (too impersonal!),
but is it possible for an outside caller to dial an internal extension (eg
201) after asterisk answers the call, but before someone in the incoming
call ring group has answered?
Thanks,
Simon
Hi. Has anyone written anything that can take CDR output and calculate
traffic intensity?
We're interested in figuring out the maximum number of simultaneous
calls we were handling for various phone numbers / services.
Thanks...
___
--Bandwidth and
Hi,
The card is detected as the following below...
lspci -vb
01:01.0 Network controller: Tiger Jet Network Inc.
Intel 537
Subsystem: Unknown device a159:0001
Flags: bus master, medium devsel, latency 32,
IRQ 10
I/O ports at b800
Memory at ff8ff000 (32-bit,
Hi,
The card is detected as the following below...
lspci -vb
01:01.0 Network controller: Tiger Jet Network Inc.
Intel 537
Subsystem: Unknown device a159:0001
Flags: bus master, medium devsel, latency 32,
IRQ 10
I/O ports at b800
Memory at ff8ff000 (32-bit,
Hi,
Can you please give me some details about the link
you have sent? I am not aware of what it does?
[http://tzafrir.org.il/genzaptelconf]
Regards,
Somesh S. Shanbhag
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, Sep 23, 2005 at 06:22:06AM -0700, somesh s
wrote:
Hi Steve,
This
On Sun, 2005-09-25 at 11:58 -0300, Antonio José dos Santos Brandão
wrote:
I`m trying a skype usb gateway pluged to a asterisk FXO port.
It's working well, but have to leave a winxp machine online.
http://www.skyvoice.com.br/
But, http://www.voip-info.org/wiki-Skype+Gateways looks better,
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