Greetings !
I am looking into the TDMoE functionality of the Zapata drivers and * and i
am kind of confused.
Lets say i have 2 linux boxes, one has * running but no fxs/fxo hardware the
other has a card (for example an x100p) but does not have * installed.
If i just want to use the card (no *
What exactly doesn't work?
Could you just paste the Asterisk debug output so that we could figure
out which part has the problem.
Maybe the forwarded-iax prompt does not exist, and it simply cannot
play it... :)
Regards,
Hohenzolern
Matthias Fechner wrote:
Hi,
I have configured asterisk
Is anybody using XEN in conjunction with Asterisk and/or Openser?
I would like to get some info about such an environment and experience
reports.
Before I had access to a spare machine, I ran it under xen and it worked
well enough to do the testing I wanted to do. I don't know that the
On 6/15/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I get annoyed Stephen when Digium goes around calling Asterisk 'enterprise
grade', which in my opinion it really isn't. I'd consider distributed ACD
queues to be a requirement for an enterprise grade product, but it's becoming
apparent
Leif Madsen wrote:
On 6/15/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I get annoyed Stephen when Digium goes around calling Asterisk
'enterprise grade', which in my opinion it really isn't. I'd consider
distributed ACD queues to be a requirement for an enterprise grade
product, but it's
Kevin Smith wrote:
It is making me lean that way, because other phones (same settings)
are using the AC adapters in another office. The ones on the adapter
have not been having this problem, but they don't use the phone much
so they may have never noticed if it did.
If you go into the ftp
- C F [EMAIL PROTECTED] wrote:
I like the TC400P card, how many T1s will that take? or is it just a
Daughter card on the TE4xx ? How many channels can it transcode?
Neither. It's a separate device, entirely unrelated to any TDM cards (which
means it can be used for any type of channel, not
- Grady Neely [EMAIL PROTECTED] wrote:
Can Asterisk emulate this INVITE Configuration? Can it send a tel URI
INVITE?
No, there is not any support in Asterisk for sending tel: URIs.
--
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.
___
On Sun, 2006-06-25 at 05:20 -0400, Leif Madsen wrote:
On 6/15/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I get annoyed Stephen when Digium goes around calling Asterisk 'enterprise
grade', which in my opinion it really isn't. I'd consider distributed ACD
queues to be a requirement for
Just a reminder that AstriCon London opens tomorrow at 8:00 AM at the
ExCeL Center in Docklands, London. If you're in London, please
consider joining us. We have one day and full conference tickets
available on the web site.
Hope to see you there!
Steve
--
Steven Sokol
AstriCon 2006:
[EMAIL PROTECTED] wrote:
- C F [EMAIL PROTECTED] wrote:
I like the TC400P card, how many T1s will that take? or is it just a
Daughter card on the TE4xx ? How many channels can it transcode?
Neither. It's a separate device, entirely unrelated to any TDM cards
(which means it can be used
Hi all
Testing the Gizmo project with asterisk led me to take a closer look
at how the communication works between the Gizmo and asterisk, and
they seem to pass all SIP and RTP traffic through their own
servers... See http://karlsbakk.net/asterisk/gizmo-project.php for
details
best
Kevin P. Fleming wrote:
- C F [EMAIL PROTECTED] wrote:
I like the TC400P card, how many T1s will that take? or is it just a
Daughter card on the TE4xx ? How many channels can it transcode?
Neither. It's a separate device, entirely unrelated to any TDM cards (which
means it can be
If you converted a wav file to gsm in the sounds directory, you have to
delete the original wav file. Not that this is the issue in your case
but just something I have run across.
Thanks,
Steve Totaro
Tigran Kocharyan wrote:
What exactly doesn't work?
Could you just paste the Asterisk debug
JP Carballo wrote:
Ronald Wiplinger wrote:
If a user calls and hangs up before the destination party rings, than
the in-use flag remains set! This is one case, but maybe there are
many other cases.
I have created a number the user can dial to reset this flag.
However, that is written in the
Hello,
Could anyone help me to figure out the following questions, please:
1. Whenever there is an incoming DTMF signal on the Zap channel, where
does the processing actually take place: In Asterisk?; or in Zaptel Drivers?
2. I'm having a problem of double (or sometimes tripple) detection of a
Hi Tigran and Steve,
Steve Totaro wrote:
If you converted a wav file to gsm in the sounds directory, you have to
delete the original wav file. Not that this is the issue in your case
but just something I have run across.
thx a lot, there was really a problem with the sound file. I have
On 6/24/06, Ronald Wiplinger [EMAIL PROTECTED] wrote:
If a user calls and hangs up before the destination party rings, than
the in-use flag remains set! This is one case, but maybe there are many
other cases.
I have created a number the user can dial to reset this flag. However,
that is written
I believe all three Aastra phones(9112, 9133i, 480i)have exactly the same
handsets and speakerphone hardware which is THE most
important thing. After that it just depends on what
additional features you want. They are ALL solid business
phones IMHO and Aastra's support of the Asterisk
On Jun 25, 2006, at 1:55 AM, Stelios Koroneos wrote:
Greetings !
I am looking into the TDMoE functionality of the Zapata drivers and *
and i
am kind of confused.
Lets say i have 2 linux boxes, one has * running but no fxs/fxo
hardware the
other has a card (for example an x100p) but does
I asked about a similar application a few weeks back. This is sometimes
referred
to as campusing since you are basically going to make the two systems sharing
their resources appear to be one system. From what I understand, you have
to have both boxes running Asterisk. I am pretty sure that
On Jun 25, 2006, at 3:29 AM, Patrick wrote:
On Sun, 2006-06-25 at 05:20 -0400, Leif Madsen wrote:
On 6/15/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I get annoyed Stephen when Digium goes around calling Asterisk
'enterprise grade', which in my opinion it really isn't. I'd
consider
Hello,
I got Quintum A800 with the P5-2-1 firmware. I configure my asterisk trunk
as followed:
[SIP_BD1]
type=peer
qualify=yes
host=192.168.0.254
disallow=all
context=from-pstn
allow=h723
and inside the quantum I change the config sip useragent to 5060. Up to this
part if I run sip show peers,
I'm not familiar with Quintum, but what codec do you mean at the allow= line
in sip.conf
with h723?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Freddy Setiawan
Sent: Sunday, June 25, 2006 8:37 PM
To: asterisk-users@lists.digium.com
Subject:
Hi,(I tried to post this message a week ago but I don't think it could reach the list. Please forgive me if you already received it).I would like to develop my first FastAGI script.I would like to test
it independently from Asterisk for the sake of simplicity.Which linux
(or cygwin) tool is the
Freddy Setiawan wrote:
Hello,
I got Quintum A800 with the P5-2-1 firmware. I configure my asterisk trunk
as followed:
[SIP_BD1]
type=peer
qualify=yes
host=192.168.0.254
disallow=all
context=from-pstn
allow=h723
and inside the quantum I change the config sip useragent to 5060. Up to this
part
I have a TDM400 card with 3x FXO and 1x FXS ports on it.
At the moment I'd prefer (till I can get it working more reliable with
iaxmodem), for a faxmodem to answer one of the lines instead of the
linecard.
I've tried changing the context of that line so that the exten = s does
nothing, but that
Thomas Kenyon wrote:
I have a TDM400 card with 3x FXO and 1x FXS ports on it.
At the moment I'd prefer (till I can get it working more reliable with
iaxmodem), for a faxmodem to answer one of the lines instead of the
linecard.
I've tried changing the context of that line so that the exten = s
Ronald Wiplinger wrote:
ok,
How do I check if a particular channel is up?
(Wasn't that what I asked above anyway)
Try this.
Using show channels concise, retrieve the channel names, then loop
through each channel using show channel channel_name.
Get the channel's UniqueID then compare
Steve Totaro wrote:
Thomas Kenyon wrote:
I have a TDM400 card with 3x FXO and 1x FXS ports on it.
At the moment I'd prefer (till I can get it working more reliable with
iaxmodem), for a faxmodem to answer one of the lines instead of the
linecard.
I've tried changing the context of that
Maxim Vexler wrote:
I would be interesed in your the applications you use in your dialplan
to reset this flag.
Could you please post here the relevent part of your extensions.conf ?
http://lists.digium.com/pipermail/asterisk-users/2006-April/146947.html
--
JP Carballo
In the quintum
also check you have a codec profile:
Example (below
has alaw and G729 configure in the codec profile):
CodecProfile-default
:
name
: (Not Set)
name
VoiceCodecAttached[1] : VoiceCodec-1
VoiceCodecAttached[2] : VoiceCodec-2
VoiceCodecAttached[3..8] :
Thomas Kenyon wrote:
Steve Totaro wrote:
Thomas Kenyon wrote:
2. No, I don't think so, I think echocancelwhenbridged=no might be
the solution there.
I hope so, that fixed the problems bridging between channels whilst
using a PDQ, I didn't have time when I was there to see
Hello Robert,
Ser, Asterisk, Mysql and Freeradius are working fine ,
I'm feeling tired with CDRRTool .
I will use an other billing system.
Thanks for your answer.
Regards
Harry
--- Robert Zorop [EMAIL PROTECTED] a écrit :
HI, i've got a working config of ser 0.9.6,
freeradius, MySQL, and
Thomas Kenyon wrote:
Thomas Kenyon wrote:
Steve Totaro wrote:
Thomas Kenyon wrote:
2. No, I don't think so, I think echocancelwhenbridged=no might be
the solution there.
I hope so, that fixed the problems bridging between channels whilst
using a PDQ, I didn't
Thomas Kenyon wrote:
Steve Totaro wrote:
Thomas Kenyon wrote:
I have a TDM400 card with 3x FXO and 1x FXS ports on it.
At the moment I'd prefer (till I can get it working more reliable with
iaxmodem), for a faxmodem to answer one of the lines instead of the
linecard.
I've tried
A2Billing V1.2.1 released today.
Place to go : http://asterisk2billing.org
Key new features :
* full rewriting on the web interface, new PHP Framework OO class
(code more structured centralized)
* CallBack : Web callback from customer interface, ANI callback, DID callback
* Gettext Multi
Hi Ronald,
If a user calls and hangs up before the destination party rings, than
the in-use flag remains set! This is one case, but maybe there are many
other cases.
You should install php-pcntl (or compile php to add support for
process control functions). The inuse problem will be fixed
Replying to myself... I was thinking on a2billing, not astcc, so
php-pcntl will make no difference.
The problem might be the same anyways. Asterisk now sends a HUP signal
to every agi script when it detects a hangup. If the script exits when
receiving the signal, it will not handle the clean up
Asterisk handling My Skype Calls
This is for me, once more, Asterisk as the Future of Telephony.
Today I've integrated my Skype Account as SIP extension in my * Box.
This has been possible using Uplink Skype to SIP Adapter, available
for free at http://www.nch.com.au/skypetosip/index.html .
Hi List,
Is there a way to tell asterisk to only accept SIP streams from the same
IP address that is used for signaling?
Thanks,
Jean-Michel.
--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
Hi Marco,
Marco Mouta wrote:
Please feel free to contact me if you have more ideas to improve this
solution, currently i didn't test more than one simultaneous calls
incoming and outgoing through Skype.
get it running on unix so you can run it on the asterisk server.
Best regards,
Matthias
I've been able to use RxFax directly to receive faxes over iax from
my DID/PSTN provider but I can't seem to get NVDetectFax to work. I
set the timeout to something ridiculously long, like 10 seconds, send
a fax with my laptop fax modem, but it never gets punted to the fax
extension,
Thanks. Gonna try today.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Neill Wilkinson
Sent: Monday, June 26, 2006 3:54
AM
To:
asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] FW:
Asterisk Quintum A800 SIP Mode
In the quintum also check
you
I tried to change the codec to ulaw but still
cannot do anything.
I got this on my Asterisk box:
-
Found RTP audio format 0
Peer audio RTP is at port
192.168.0.254:10240
Found description format pcmu
Are you using PoE? I had two PoE IP501s on long LAN runs, and the
phones would periodically reboot. I plugged an AC adapter into the
phone (with PoE) and everything is fine.
Kevin Smith wrote:
Hey everyone,
I know this isn't a direct Asterisk issue, but some of you may know this
answer.
On 6/24/06, Patrick [EMAIL PROTECTED] wrote:
On Fri, 2006-06-23 at 14:58 -0400, Matt Florell wrote:
P4 3.2GHz
HT enabled
1MB L3 cache
2GB RAM
Asterisk 1.2
meetme participants were Zap or IAX, some rooms recording
Matt,
A bit off-topic but I noticed you have HT enabled. Since you may be
Still awfully pricey for home use and the styling is not there for a
bedroom or many other areas of a modern home. What we need is a wireless
sip phone modeled like the panasonic or uniden which allow multiple
extension off of one base. The base would connect to the internet. The
other problem is
I have a issue trying to understand why Asterisk-PBX, when a SNOM
(320 or 360) successfully redirects/diverts a call when it is a local
extension, but fails when you enter external number.
Both the local extension dial and external extension dial are within
the same context [from-sip] and
Does anyone know of any startups using Asterisk? What about established
companies? Ones that are hiring would be nice :)
Doug.
___
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To UNSUBSCRIBE or update
Douglas Garstang wrote:
Does anyone know of any startups using Asterisk? What about established
companies? Ones that are hiring would be nice :)
Doug.
___
On Jun 25, 2006, at 4:11 PM, Jean-Michel Hiver wrote:
Hi List,
Is there a way to tell asterisk to only accept SIP streams from the
same IP address that is used for signaling?
You mean like setting reinvite and canreinvite to no in your
extensions.conf? This forces asterisk to stay in the
Title: SIP Channel hangup problem with re-INVITE enabled - ugrent
Does anyone on this list has idea?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hoa Thai
DuySent: Thursday, June 22, 2006 2:50 PMTo: 'Asterisk
Users Mailing List - Non-Commercial Discussion'Cc:
Paul,
D'oh. The fact I left Sydney 5 years ago for the US might be a teeny
complication. :P
Doug.
-Original Message-
From: Paul Hales [mailto:[EMAIL PROTECTED]
Sent: Sun 6/25/2006 11:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
To all Employable Asterisk Professionals,
We are very pleased to announce the unveiling of the newest incarnation
of the popular, OpenSource VOIP jobs forum at http://www.asterisk-jobs.com.
We at Asterisk-Jobs.com appologize for the inactivity for the past while.
It had come to our attention
Well, I hope some more jobs get posted. I took a look tonight, and there was 2
there.
-Original Message-
From: Matt Gibson [mailto:[EMAIL PROTECTED]
Sent: Sun 6/25/2006 11:25 PM
To: asterisk-users@lists.digium.com
Cc:
Subject:
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