On Wednesday 12 Oct 2005 14:54, Tom Rymes wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
trixter http://www.0xdecafbad.com
Sent: Tuesday, October 11, 2005 5:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Cory Andrews wrote:
Yeah I should have picked up on that, single PCI Riser in this one, so 1
card. I don't know of any 1U solution out there that would give you 3
PCI slots to work with, I think you'll have to go to a 2U at least to
achieve this.
I saw the Dell PowerEdge 1850 has 2 PCI-X on
hy all
i need a suggesion on what hardware should i use for
the following case study
i have five offices each will be having 35 to 45
extensions. if i will be using voip fones for those
extensions( either it is iax or sip ) which one will
be better and cheaper what should i use. all the five
I recommend checking the following site...
www.voip-info.org
Lots of info for you there...
By VoIP phones, I think you are meaning soft phones which are software
based.
You will need a headset for the PC that runs the software phone.
Usually Logitech or Plantronics at about $50 a headset.
If
ishtiaq ahmed wrote:
hy all
i need a suggesion on what hardware should i use for
the following case study
i have five offices each will be having 35 to 45
extensions. if i will be using voip fones for those
extensions( either it is iax or sip ) which one will
be better and cheaper what
Hello
We have discovered a problem with DTMF on Asterisk.
We have a setup with a T1 from PSTN going into an Asterisk box, and
then out again on T1 and into a normal PBX (EADS)
We use it to record all calls going to/from the PBX.
The problem is that when we record the calls (with MONITOR
Thanks for the confirmation!
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On 10/12/05, Mir [EMAIL PROTECTED] wrote:
Hello
We have discovered a problem with DTMF on Asterisk.
We have a setup with a T1 from PSTN going into an Asterisk box, and
then out again on T1 and into a normal PBX (EADS)
We use it to record all calls going to/from the PBX.
The problem is
We use the SmartNodes SN1400 and SN2300 as ISDN Gateways in our customer
Asterisk installations and are really happy with them. They run very stable
and you can configure nearly everything. Support from INALP is also great.
With the interface cards for the SmartNode 2300 you should be able to
I try to integrate my old PBX Meridian and Asterisk througth a PRI T1 (I'm
gonna use only the asterisk voicemail system) but i don't know how to
integrate the MWI protocol between Asterisk Voicemail and my Nortel
meridian.
Anyone know what i have to do for that.?
Any idea is appreciated.
I've been wanting to do exactly the same thing, but I believe it's beyond my coding skills.
I think we need a function similar to the SirrixMWI. Some initial
code for MWI exists in libpri, but nothing in the rest of asterisk
calls those functions yet.
On 10/12/05, kritikus Araklidas [EMAIL
TKS buddy if i find o develop myself something regarding that i told you.
Cristian.
From: Gary Reuter [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
I have set up a asterisk-server behind NAT and peers to another asterisk
and uses that one for outgoing calls. I have som clients on my asterisk
and they could register to it well over internet. Not a problem. But when
they try to call me the asterisk-server tells me this:
Oct 12 23:21:38
Bob Goddard wrote:
On Wednesday 12 Oct 2005 14:53, Tom Rymes wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bob Goddard
Sent: Tuesday, October 11, 2005 6:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Either Permissions on the directory are incorrect or you have no unavail.wav file.On 10/11/05, Andy Goss
[EMAIL PROTECTED] wrote:I migrated to a new version of the voip connection vs1 server software
and I am now getting these errors when I try to call my voicemail.Anythoughts?The files are
On Wed, 2005-10-12 at 16:48 -0500, Tim Litwiller wrote:
See IAXModem above for the soft DSP.
There is very little info on the sf.net page regarding its
capabilities ...
Does it only do fax or does it do other data communications?
What fax protocols are supported?
Does the destination
I noticed that using a TDM04B and only having 3 analog lines connected
at this time (4th is coming)
I was using ChanIsAvail(Zap/4Zap/3Zap/2Zap1) which gives me an
available line no problem.
However Zap/3 did not actually have a line connected at this time. Yet
asterisk still gave me it as
Hi,
I was wondering if I could use Asterisk logo in my PBX system which I
want to introduce in my local market. Does anyone know if I must fill
some legal issues which let me use this logo.
Best regards
Andrew
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trixter http://www.0xdecafbad.com wrote:
There is very little info on the sf.net page regarding its
capabilities ...
Right now that's intentional. I still consider it developer-grade
code. That said, I do use it on small production usage, and it's fine
there with a few known issues
On Oct 12, 2005, at 3:44 PM, Bob Goddard wrote:
Asterisk's faxing capabilities are not nearly as advanced, stable, or
easy to set up as HylaFAX. Also, there seem to be many problems with
frame slipping and the like that screw up faxing over Digium
cards, and
maybe others as well.
Does
On Oct 12, 2005, at 11:26 AM, Lee Howard wrote:
Tom Rymes wrote:
(I would like to be able to receive faxes reliably
over our PRI)
Until then, however, I still recommend HylaFAX.
If your PRI comes in to a TE405P or somesuch then you can pass fax
DIDs out through another port on the
Digium has stated that you need their permission to use the logos.
However, I was under the impression that if you attributed the
copyright to them that you would not need their permission. Of
course, I am not a copyright lawyer ... (IANACL?)
In other words: Asterisk and the Asterisk logo
Can multiple soft phones (running on separate computers) be used
simultaneously from the same outside IP address?
Yep, should work fine.
Consider how any one webserver can handle multiple http requests to port 80.
Or consider when what happens when multiple users behind the same nat
firewall
On Wed, 2005-10-12 at 08:59 +, Andrew Nowrot wrote:
Hi,
I was wondering if I could use Asterisk logo in my PBX system which I
want to introduce in my local market. Does anyone know if I must fill
some legal issues which let me use this logo.
Best regards
digium is the owner of that,
My question is, will this support more than 1 simultaneous connection from
the same outside IP address, or will only one soft phone function?
or, put another way:
Can multiple soft phones (running on separate computers) be used
simultaneously from the same outside IP address?
I've
Does everything with AstCC work properly under Asterisk 1.2?
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On Wed, 2005-10-12 at 18:18 -0400, Tom Rymes wrote:
On Oct 12, 2005, at 11:26 AM, Lee Howard wrote:
If your PRI comes in to a TE405P or somesuch then you can pass fax
DIDs out through another port on the TE405P and out to a T1
faxmodem (such as a Patton 2977) or a T1 channel bank and
Tom Rymes wrote:
On Oct 12, 2005, at 11:26 AM, Lee Howard wrote:
Tom Rymes wrote:
(I would like to be able to receive faxes reliably
over our PRI)
Until then, however, I still recommend HylaFAX.
If your PRI comes in to a TE405P or somesuch then you can pass fax
DIDs out through
On Oct 12, 2005, at 1:30 PM, Pedro Nunes wrote:
Hi there,
Does anyone know how to setup an overflow queue? When a call rings
on the queue A, if all agents were busy, the call goes to the queue B.
If all agents in queue B were busy, then the call stays on both
queues until somebody answers
trixter http://www.0xdecafbad.com wrote:
Cant iaxmodem work by having asterisk bridge the pri channel as needed
(did based routing perhaps) and then have hylafax use iaxmodem as the
modem it uses. That should result in no additional hardware, which
means testing can happen with little cost to
On Oct 12, 2005, at 6:33 PM, trixter http://www.0xdecafbad.com wrote:
On Wed, 2005-10-12 at 18:18 -0400, Tom Rymes wrote:
On Oct 12, 2005, at 11:26 AM, Lee Howard wrote:
If your PRI comes in to a TE405P or somesuch then you can pass fax
DIDs out through another port on the TE405P and out
Is there a place where all the parameters are documented ?
In example (my example!) I would like to know the meaning of a lot of
parameter that can be used in sip.conf,
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
How did I found this ?
I'll second that... good reading so far, have not finished it though.
Rich
The people who have been documenting Asterisk have been working on a book
for the last few months, it has been published by O'reilly (Asterisk-The
Future of Telephony)and is just now finding
On Wed, 2005-10-12 at 18:45 -0400, Tom Rymes wrote:
This is true, but:
1.) Lee has stated that IAXModem is still Developer-grade code.
2.) I don't have a spare PRI for testing, and our phone system is far
too mission critical for me to go mucking about with it and trying
this out
On Wed, 2005-10-12 at 09:41 -0400, Time Bandit wrote:
Is there a place where all the parameters are documented ?
In example (my example!) I would like to know the meaning of a lot of
parameter that can be used in sip.conf,
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
How did
On Oct 12, 2005, at 6:58 PM, trixter http://www.0xdecafbad.com wrote:
On Wed, 2005-10-12 at 18:45 -0400, Tom Rymes wrote:
This is true, but:
1.) Lee has stated that IAXModem is still Developer-grade code.
2.) I don't have a spare PRI for testing, and our phone system is far
too mission
I noticed that using a TDM04B and only having 3 analog lines connected
at this time (4th is coming)
I was using ChanIsAvail(Zap/4Zap/3Zap/2Zap1) which gives me an
available line no problem.
However Zap/3 did not actually have a line connected at this time. Yet
asterisk still gave
On Wed, 2005-10-12 at 19:19 -0400, Tom Rymes wrote:
Would that not solve in the short term all of those issues or am I
missing something?
Well, I can redirect a DID to it, but I have no fax traffic going to
that DID, and I am still reluctant to install developer-grade code
on my
On Wed, Oct 12, 2005 at 08:43:32PM +0200, Bruno Voigt wrote:
I'm looking for a way with any asterisk-version with TE410P (cpe
EuroISDN, Q931)
for sending an INFORMATION ELEMENT KeypadFacility,
eg. *87, during a connected call to the PSTN switch.
Are there existing functions in asterisk to
Asterisk cvs-head compiled 2005-10-07 11:
Oct 12 18:35:12 NOTICE[21740]: chan_sip.c:10685 handle_request_register: Registr
ation from 'sip:[EMAIL PROTECTED]' failed for '208.5.218.28' - Not a lo
cal SIP domain
The sip phone is a Cisco 7960 with one line defined, and registration
with * is
trixter http://www.0xdecafbad.com wrote:
Perhaps Lee can comment on exactly how 'development grade' it really is,
perhaps even cite some test cases where people have used it on larger
scale operations (ie larger than a home users 1-2 times a month or
less).
According to sourceforge, there
You need about 30MHz per channel. That means the Soekris can only handle part
of a T1, it will never handle a quad span.
Paul
--- Kristian Kielhofner [EMAIL PROTECTED] wrote:
Craig Guy wrote:
Has anyone on the list used a Soekris engineering PC as a TDM - Ethernet
bridge? For example
Hi,
Maybe you can record the sound file vm-five.gsm as five hour in Japanese, instead of just five.
AK
On 10/12/05, Kuniyoshi Murata [EMAIL PROTECTED] wrote:
Dear Asterisk Users,I'm a Japanese and now configuring Voicemail.Now I need to modify the way cmd VoicemailMain works to fix language
Andy Kuo writes:
Hi,
Maybe you can record the sound file vm-five.gsm as five hour in
Japanese, instead of just five.
AK
I don't think you can do that.
Because that vm-five.gsm can be used as message number also (e.g. message FIVE)
--
Kuniyoshi
On Wed, 2005-10-12 at 17:46 -0700, Paul Mahler wrote:
You need about 30MHz per channel. That means the Soekris can only handle part
of a T1, it will never handle a quad span.
Paul
How was that determined?
I have a problem with a plain number like that, which may have been
taken into
On Thu, 2005-10-13 at 10:08 +0900, Kuniyoshi Murata wrote:
Andy Kuo writes:
Hi,
Maybe you can record the sound file vm-five.gsm as five hour in
Japanese, instead of just five.
AK
I don't think you can do that.
Because that vm-five.gsm can be used as message number also (e.g.
Lee Howard [EMAIL PROTECTED] wrote:
Yes I have a Patton 2977 (driven by HylaFAX), connected via crossover to
one port on a TE405P (driven by Asterisk) which has another port connected
to the T1 from the telco. Asterisk bridges the two for sending and
receiving. Receiving is wonderful.
Hello,
I've released an Asterisk application under the terms of the GNU GPL. You
may find it here:
http://psg.com/~begg/projects/
A short exerpt from the README:
--
Broadcast is an Asterisk (http://www.asterisk.org) application which you
may use to send a generic message over TCP/IP to any
Darren Nickerson wrote:
We prefer the Eicon Diva server and Brooktrout TR1034 boards, which
are known to work well with HylaFAX since we've had our share of
headaches with the 2977's.
Well, part of my preference for the 2977s involves my strong dislike for
the way that the Diva Servers and
I have also been looking for a way to customize voicemail (I want to add a
pause feature and change the promps). I have come to the same conclusions
as to where to do it, but have not yet created a solution. I have found
this posting/forum which gives insight into modifying the
Lee Howard [EMAIL PROTECTED]
Well, part of my preference for the 2977s involves my strong dislike for
the way that the Diva Servers and BrookTrouts do things. It's enough of a
dislike to get me over the learning curve of how to properly set up the
2977s for HylaFAX use.
I agree, there's a
I want to give the users the announcements as they subscribed to. The
announcements should be in English, Chinese, Cantonese, according to
their phone number. How can I do that? I can hardly make for each number
a different context!!!
bye
Ronald Wiplinger
Does anyone know of a good solution to create a secure
(encrypted) connection from a pocketpc (IPAQ 6515 in my case) to an asterisk
server?
Thanks
Peter Kellner
http://PeterKellner.net
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- Original Message -
From: MvPhone
To: Asterisk Users Mailing List -
Darren Nickerson wrote:
I'm curious though, in an earlier message you wrote:
Sending is also quite good, however, there are some quirks with the
Patton
firmware which need to be resolved for me to be completely delighted.
Are these issues you refer to just cosmetic 'nice to haves' or do
Hi everyone!
I've been working on setting up an Asterisk server and my two Digium
cards I ordered will arrive tommorow, so I'm excited to plug some 'real'
old-school lines into it.
But tonight I've been testing with some of our staff around the world,
and while handing off 'real' (PSTN - over
On Thu, 2005-10-13 at 12:17 +0800, Ronald Wiplinger wrote:
I want to give the users the announcements as they subscribed to. The
announcements should be in English, Chinese, Cantonese, according to
their phone number. How can I do that? I can hardly make for each number
a different
modprobe zaptel is successful. When I do lsmod zaptel
is loaded.
Regards,
Somesh S. Shanbhag
--- Lyle Giese [EMAIL PROTECTED] wrote:
I have not seen the output of modprob zaptel in this
thread, which has
to take place before loading the other kernel
drivers.
Lyle
so
mesh s wrote:
Hi everyone!
I've been working on setting up an Asterisk server and my two Digium
cards I ordered will arrive tommorow, so I'm excited to plug some 'real'
old-school lines into it.
But tonight I've been testing with some of our staff around the world,
and while handing off 'real' (PSTN - over
Hi,
We have a Data/Voice service supplied through an
integrated T1.
Does anyone know if Digium T1 card will support the
splitting of the Voice and Data?
Regards.
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