Hello!
I m reading all sorts of descriptions of FastAGI() on the internet but
I m finding no information how asterisk signals the remote process
that the connection has been hungup. In an AGI script you receive a
SIGHUP. What happens with a FastAGI connection?
Regards
Hello
I need to hook up someone's remote PC onto our Asterisk server over
the Net. There are firewalls on each side, so I figured it's time to
give IAX a try, and see if it's less of a pain to use than SIP. And
since IAX hardphones are pretty are, I guess I'll go softphone.
Apparently,
Hi,
I've been trying to configure my extensions.conf and sip.conf for two days
now and I'm pretty sure it's just a small typo or anything I can't find by
myself.
My setup:
- Asterisk connected via Fritz! PCI Card to a HiPath 3500 (2 channels)
- Callcentric.com SIP channel to dial out to foreign
Hello,
This is just a warning, that a snom phone tries to subscribe an
extension which has no hint entry. You should try to find the snom which
has set up the subscription which wasn´t found. You should search in the
snom webif in the function keys for the function nebenstelle or
extension in
Hi,
according to the description of Pickup() on page
http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
I can use this command to pickup a call at a certain extensions.
When I try this with e.g.
exten = *8200,1,Pickup(200)
Asterisk tells me that the highest value for the Pickup command
Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Feb 02, 2008 at 09:46:42AM +0100, Johansson Olle E wrote:
Friends,
I'm having severe problems with zaptel timers on Intel Dual Core
systems with SMP code enabled. Ztdummy, zaptel connected to Digium TDM
or PRI cards - all ends up with
On Tue, Feb 05, 2008 at 09:58:54AM +, Tony Mountifield wrote:
Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Feb 02, 2008 at 09:46:42AM +0100, Johansson Olle E wrote:
Friends,
I'm having severe problems with zaptel timers on Intel Dual Core
systems with SMP code enabled.
On Tue, 5 Feb 2008, Sebastian Pape wrote:
Hi,
I've been trying to configure my extensions.conf and sip.conf for two days
now and I'm pretty sure it's just a small typo or anything I can't find by
myself.
My setup:
- Asterisk connected via Fritz! PCI Card to a HiPath 3500 (2 channels)
-
On Tue, 5 Feb 2008, Vincent wrote:
Hello
I need to hook up someone's remote PC onto our Asterisk server over
the Net. There are firewalls on each side, so I figured it's time to
give IAX a try, and see if it's less of a pain to use than SIP. And
since IAX hardphones are pretty are, I
Sorry,
I tried to use underscore(s) before the variable names, but without any
success.
H234m_gw is a functionality which we use for video calling on asterisk.
(http://sip.fontventa.com/)
--
Arjan Kroon
Mobillion BV
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Jared was talking about a decent IAX hardphone on this list a week or so back,
I don't recall the make.
If you use IAX, all you need to do is :
1) set your local firewall to forward udp 4569 to asterisk.
(optionally filtering by from IP address if your user has a
fixed IP address or known
This is a part of our programma.
[begin]
exten = s,1, h324m_gw([EMAIL PROTECTED])
[video]
exten = s,1,h324m_gw_answer()
exten = s,2,Wait(3)
exten = s,3,Goto(intro,s,1)
[intro]
exten = s,1,mp4play(intro.3gp)
exten = #,n,Goto(einde,s,1)
[einde]
exten = s,n, Hangup()
When I use this dialplan
Hi,
try use Dial with G parameter, and bridge these to extensions with meetme.
But the problem is that, I don't know how to close conference, when one
hangups...
2007/11/2, Asterisk [EMAIL PROTECTED]:
Hi there,
I'm trying to bridge 2 SIP channels together via AGI script. The AGI
script is
Hi List,
I have this running, but after I park a call it will not announce where it
is at, it's like you have to call another application just to say where it
is parked at. I have tried a second priority option for the same extension
with that ValetParkList but it seems once ValetParkCall
I have been using the D-Link DPH-540 wireless VOIP handset, and I really
like this phone. We had tried the UStarcomm phone, but the phone is used in
a noisy environment and the volume wasn't loud enough. The problem with
the D-Link phone is the Li-ion battery needs to be replaced and D-Link
On 2/5/08, Arjan Kroon | Mobillion [EMAIL PROTECTED] wrote:
This is a part of our programma.
[begin]
exten = s,1, h324m_gw([EMAIL PROTECTED])
[video]
exten = s,1,h324m_gw_answer()
exten = s,2,Wait(3)
exten = s,3,Goto(intro,s,1)
[intro]
exten = s,1,mp4play(intro.3gp)
exten =
Marc Charbonneau wrote:
- shameless plugMy MediaX softphone :
http://www.marccharbonneau.com/asterisk/mediaxphone.php/shameless
plug
Marc, does your client play nicely with Vista? We've been having some
problems with softphones that work fine in XP, but choke in Vista.
--
Alan Williamson
Are other clients I should know about?
http://www.zoiper.com/
http://www.counterpath.com/
Add to that list
- Mozphone (http://mozphone.mozdev.org/) that can be installed in Firefox
-Kiax : http://sourceforge.net/projects/kiax
- shameless plugMy MediaX softphone :
We're using Pirelli DPL10's and nokia N95's with cisco aironet access
points and both phones are quite happy roaming around the building (6
access points) during calls - the nokias seem to have better signal
strength and audio quality than the pirelli's though.
Geraint
SIP wrote:
Brian J.
Hello,
I'm doing some research concerning iax encryption, I haven't find any
clients (softphones or hardphones) which implement so I have not tested
it yet.
There was also this message on asterisk-security mailing list
http://archives.free.net.ph/message/20070507.101933.222987b2.en.html
which got
On Tue, 2008-02-05 at 14:37 +, [EMAIL PROTECTED] wrote:
Hi,
I use Linksys WIP 330 and the sound is good, talk time with full battery go
up to 2 hours, I'm happy with.
Ahhh. OP wanted to know about wirelessly networked phones. Interesting
as they are (and expensive -- the WIP-330
Hi,
I use Linksys WIP 330 and the sound is good, talk time with full battery go up
to 2 hours, I'm happy with.
Best regards,
Chris Hariga
Sent from my BlackBerry® wireless device
-Original Message-
From: marvin horst [EMAIL PROTECTED]
Date: Tue, 5 Feb 2008 08:52:44
Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Feb 05, 2008 at 09:58:54AM +, Tony Mountifield wrote:
Specifically, what kernel version?
As I understand from an IRC chat - CentOS 5 - 2.6.18-something.
Hmm, ok.
If you are using 2.6.9 (e.g. Centos 4),
then by default when you build
Brian J. Murrell wrote:
On Tue, 2008-02-05 at 14:37 +, [EMAIL PROTECTED] wrote:
Hi,
I use Linksys WIP 330 and the sound is good, talk time with full battery go
up to 2 hours, I'm happy with.
Ahhh. OP wanted to know about wirelessly networked phones. Interesting
as they are
On Tue, 5 Feb 2008, Tony Mountifield wrote:
Tzafrir Cohen [EMAIL PROTECTED] wrote:
So that point is mute.
moot, not mute.
moot, not mute.
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice:
However, in general, none of the true Wi-Fi phones we've tested other
than the Nokia E series have been worth mucking with. Dropping off APs,
poor NAT capability, low battery life, troublesome configurations,
random weirdness -- these seem to abound in the world of wi-fi SIP. This
is why
Stefan Reuter wrote:
Lee Jenkins wrote:
I thought that the OP was asking for something to perl what Asterisk-Java
does
for java coders. I would definitely consider Asterisk-Java to be a
framework,
though not so much with PasAGI which is more of an class object wrapper
around
AGI
We are maxed out on our legacy PBX, and the question is the process of
migrating to a new * system from the legacy. We current have 36 FXO
lines coming into our site, and the usage on these lines indicates we can
spare a few of them to launch the * server, and then move additional lines
over as
Asterisk 1.6 includes a new feature that allows using AMI as a transport
for AGI commands, there abstraction becomes even more important.
For Asterisk-Java I am currently adding support for that in a way that
allows the developer to run the same AGI code either through FastAGI
or AMI
Hi,
I have asterisk installed in the xen virtual server.
I installed zaptel 1.4.2.1 and patched it to have ztxen module.
I loaded ztxen module but when I try to invoke or call to my meetme
application
I get the following warning and negative result of connecting to conference:
[Feb 5 17:46:13]
In article [EMAIL PROTECTED],
Steve Edwards [EMAIL PROTECTED] wrote:
On Tue, 5 Feb 2008, Tony Mountifield wrote:
Tzafrir Cohen [EMAIL PROTECTED] wrote:
So that point is mute.
moot, not mute.
moot, not mute.
Now that point really *is* moot (i.e. open to debate, which is the true
Ok, so I've asked this question before, and didn't get an answer.
So here I go again!
Asterisk 1.4 has some channel variables that you can inspect after a call is
complete that will give you QoS metrics. Stuff like average round trip time,
etc.
Since there's only one set of variables, and
I have found several references to this problem, but never a solution. I
have fixed it before, but it was always by accident...
Feb 5 13:27:39 NOTICE[12924]: channel.c:1904 ast_read: Dropping
incompatible voice frame on Local/[EMAIL PROTECTED],2 of
format ulaw since our native format has
Thomas Kenyon wrote:
The server that I will need to get this running on has an 82801EB/ER
(ICH5/ICH5R) AC'97 sound controller (and no expansion space left to put
another card in).
Just a suggestion, don't forget there are USB audio devices available
that work with linux, you may have an
Hi Paul,
Am Dienstag, den 05.02.2008, 10:10 +1100 schrieb Paul Hales:
With some of the phones (snom, for example) you can turn off mwi, so the
phone will only accept one call at a time. Much easier.
PaulH
Thanks for Your answer. Unfortunaly turning call waiting off is not an
option for me.
We are maxed out on our legacy PBX, and the question is the process of
migrating to a new * system from the legacy. We current have 36 FXO lines
coming into our site, and the usage on these lines indicates we can spare a
few
of them to launch the * server, and then move additional lines
Gah. So currently in 1.4, there is no method of having Asterisk accept SIP
NOTIFY from another server, and pass it on to endpoints if it matches? I
can't imagine this being that complex, but then again I'm not familiar with
the Asterisk internals. It just seems Asterisk would compare the SIP
Michael Graves wrote:
However, in general, none of the true Wi-Fi phones we've tested other
than the Nokia E series have been worth mucking with. Dropping off APs,
poor NAT capability, low battery life, troublesome configurations,
random weirdness -- these seem to abound in the world of
randulo wrote:
On Feb 4, 2008 9:34 PM, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
In my recollection, [EMAIL PROTECTED] worked when I tried it, without sip
or a colon. xxx could be anything at all. I noted this behavior back
in 2006:
Hi
I need a small PBX for use on the move. This means that outbound calls
will need to be made over the cell phone network.
Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot
then what hardware options do I have to get an outbound cellular
channel? Options need to be
I have a asterisk server. Two SIP Soft XLites are connected to the server. I am
able to make
calls from one SIP Phones to the other SIP Phones and landlines successfully.
The SIP Soft Phone on th eother side can hear my voice but I cannot hear their
voice.
They can call my local cell phone as
Ed W wrote:
Hi
I need a small PBX for use on the move. This means that outbound calls
will need to be made over the cell phone network.
Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot
then what hardware options do I have to get an outbound cellular
channel?
On Feb 5, 2008 2:32 PM, Sanjoy Rath [EMAIL PROTECTED] wrote:
The Asterisk server is a linux server. There is no firewall between the
servers. It is in a DMZ.
My bet is that it's not a *true* DMZ. You're still dealing with NAT,
and that's what's causing the one-way audio.
This topic has been
On Feb 5, 2008 2:37 PM, Drew Gibson [EMAIL PROTECTED] wrote:
How about http://www.mgamble.ca/oss/iphone_asterisk/ ?
Hah! Cool, but quite ridiculous. :-)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
I have an asterisk server. Two SIP Soft XLites are connected to the server. I
am able to make calls from one SIP Phones to the other SIP Phones and landlines
successfully. The SIP Soft Phone on th eother side can hear my voice but I
cannot hear their voice. They can call my local cell phone as
Hi Stefan,
Am Dienstag, den 05.02.2008, 10:30 +0100 schrieb Stefan Guenther:
Hi,
according to the description of Pickup() on page
http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
I can use this command to pickup a call at a certain extensions.
When I try this with e.g.
exten
Erik Anderson wrote:
On Feb 5, 2008 2:37 PM, Drew Gibson [EMAIL PROTECTED] wrote:
How about http://www.mgamble.ca/oss/iphone_asterisk/ ?
Hah! Cool, but quite ridiculous. :-)
I have a Linksys NSLU2 (Slug) at home running Asterisk (see
http://www.nslu2-linux.org/ )
It's small,
Erik Anderson wrote:
On Feb 5, 2008 2:32 PM, Sanjoy Rath [EMAIL PROTECTED] wrote:
The Asterisk server is a linux server. There is no firewall between the
servers. It is in a DMZ.
My bet is that it's not a *true* DMZ. You're still dealing with NAT,
and that's what's causing the one-way
So any recommendations for another wireless VOIP phone?
As someone else pointed out, the Siemens C450 IP (and higher models) work great!
Also, the snom m3 gets some good reviews and will be the next one I'll try out..
cheers,
stoffell
___
--
A linksys PAP2 with a Motorola Dect set is what I use for a wireless IP
phone solution. I have tried Zyxel y Linksys wifi phones, and a couple of
others, but the battery life just isn't workable on WIFI phones.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
As someone else pointed out, the Siemens C450 IP (and higher models) work
great!
I should point out that for the relatively small price difference, it's well
worth getting the S450 rather than the C460. The screen on the 's' series is
much more crisp and higher resolution. If you use the
- Original Message
From: Jason Crum [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, 5 February, 2008 7:13:12 PM
Subject: Re: [asterisk-users] External MWI question for Asterisk
Gah. So currently in 1.4,
On Tue, 5 Feb 2008 22:35:38 -, Chris Bagnall wrote:
As someone else pointed out, the Siemens C450 IP (and higher models) work
great!
I should point out that for the relatively small price difference, it's well
worth getting the S450 rather than the C460. The screen on the 's' series is
Ah, so you have a MWI service that polls the Asterisk realtime DB for the
SIP URI information for an external voicemail system? I'm guessing whatever
you're using for voicemail alerts the your MWI service (custom written for
this?)?
On Feb 5, 2008 5:36 PM, Grey Man [EMAIL PROTECTED] wrote:
Hi
2849 (20080205) __
The message was checked by ESET Smart Security.
http://www.eset.com
__ Information from ESET Smart Security, version of virus signature
database 2851 (20080205) __
The message was checked by ESET Smart Security.
http://www.eset.com
Hi list,
After recently setting up voicemail for Asterisk 1.4.14 on my Debian
etch server, I noticed that I can't delete any old voicemail messages.
The voicemail menu option Press 7 to delete this message is
available, but when I press 7 the response is always message
undeleted and the
On 00:38, Wed 06 Feb 08, Jaap Winius wrote:
Hi list,
After recently setting up voicemail for Asterisk 1.4.14 on my Debian
etch server, I noticed that I can't delete any old voicemail messages.
The voicemail menu option Press 7 to delete this message is
available, but when I press 7
Mojo with Horan Company, LLC wrote:
Thomas Kenyon wrote:
The server that I will need to get this running on has an 82801EB/ER
(ICH5/ICH5R) AC'97 sound controller (and no expansion space left to put
another card in).
Just a suggestion, don't forget there are USB audio devices available
Quoting Michiel van Baak [EMAIL PROTECTED]:
On 00:38, Wed 06 Feb 08, Jaap Winius wrote:
Hi list,
After recently setting up voicemail for Asterisk 1.4.14 on my Debian
etch server, I noticed that I can't delete any old voicemail messages.
The voicemail menu option Press 7 to delete this
- Original Message
From: Jason Crum [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, 5 February, 2008 10:59:59 PM
Subject: Re: [asterisk-users] External MWI question for Asterisk
Ah, so you have a
Well I think you need a GSM Gateway
You can find some info on cyber-telecom.net
For a cheap option you can try a CT-G1000 or CT-G2000 and then plug it in a
X100P or something similar then it would be very economical.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi,
Just wanted to let you know that we have just made our
GPL toolkit Gemeinschaft available to the public. (Finally.)
Mostly German for now - about half of the strings in the
language strings file have been translated to English.
I'm a software developer, not a marketing guy, so ...
svn co
Marc, does your client play nicely with Vista? We've been having some
problems with softphones that work fine in XP, but choke in Vista.
I don't know, never tried it since I couldn't find a machine with
enough power to run Vista decently ;)
Try it and let me know how it goes.
If it doesn't
Quoting Andy Doss [EMAIL PROTECTED]:
File permission error?
That is just my first guess. I am kind of new to Asterisk myself.
The files are all in /var/spool/asterisk/voicemail/ where the asterisk
user has read/write access to everything. Also, I see no error
messages that would indicate a
I am trying to set up Astunicall 1.4.16 with a link from Alestra in
Mexico City. I have done everything I usually do for other links in
Mexico but this one simply will not send or receive calls. I just get
Protocol error.
Anyone has any experience with R2 and Alestra?
--
Moises Silva wrote:
Asterisk 1.6 includes a new feature that allows using AMI as a transport
for AGI commands, there abstraction becomes even more important.
For Asterisk-Java I am currently adding support for that in a way that
allows the developer to run the same AGI code either through
Very Nice!
Its much more reliable than translating DSCP to COS by switch which i'm not
sure which switch does that and which one doesn't, and how they do it
considering some gray area when you translate from DSCP to COS.
On Feb 4, 2008 5:26 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Sun,
Hello All,
I have 2 new Grandstream GXV 3000 phones and want to sell them to someone
who is interested to buy. I can sell $200 per piece.
If you are interested please reply this mail.
Thanks,
Thameem
On May 8, 2007 7:25 AM, Nitesh Divecha [EMAIL PROTECTED] wrote:
Hello,
So far yes... The
Hi,
2008/2/5, Grey Man [EMAIL PROTECTED]:
- Original Message
From: Jason Crum [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, 5 February, 2008 7:13:12 PM
Subject: Re: [asterisk-users] External MWI
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