On 3/28/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
I am using autoload and I have rebooted the server. I have tried using
different files and a different location. This is getting very
frustrating.
I wish I knew what the problem was.
Not that it will help me, because I'm pretty much
Hi Carlos,
this happens to me when oppanel server is not working. Check it is running.
Giorgio
Carlos Jerónimo wrote:
HI!!!Sorry this post about FOP but it's important.
Ive installed asterisk and freepbx. the interface of FreePBX works
fine, but when i acesse FOP
(Flash Operator Panel) i get
Matt wrote:
Do you mean queue? If so, yes this is a very easy thing to do and is
document on the voip-info.org http://voip-info.org wiki under the
queues section.
Thank you and excuse me I'am a totally newbie in VoIP and tel!
I solved my problem with queue.
On 3/26/07, * Suity Zsolt*
27 mar 2007 kl. 10.48 skrev Tim Panton:
On 26 Mar 2007, at 22:32, Michael Graves wrote:
Hi All,
I've been reading about Phil Zimmermann's ZRTP encryption scheme for
SIP clients. This seems attactive but I don't use soft phones. I'm
guessing that we'd need ZRTP support in Asterisk in
order
hello I want to install Asterisk just to use in my LAN, without a analog or
digital devices.
I need to install all this packages???
Asterisk 1.2.17
Zaptel 1.2.16
Libpri 1.2.4
Addons 1.2.5
Sounds 1.2.1
thanks
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Hi,
I've just observed something a bit odd - I'm wondering if this is the
expected behaviour, a bug/feature, or something I'm doing stupid!
1st person gets into MeetMe. Nothing fancy, just:
exten = 987,1,MeetMe(400,iM)
They enter the passcode and their name, then listen to MoH. So-far so
Hello,
I would like to generate a peer-to-peer or a server/client
SIP traffic between two or more Openwrt access point, to
make some statistics about QoS. I tried some SIP traffic
generators for OpenWrt, but I didn't find nothing of
satisfactory.
Now I wonder if asterisk can help me generating
This is the expected behavior -- if the second person comes in and you
have name announcements, then the first person will hear that and
should have the sense to know not to hang up. You can have everybody
hear music till a certain person comes in, if you want.
on Wednesday 03/28/2007 Gordon
Hi ,
I am new to Asterisk community. I have some queries. Please
guide me on the following :
1)I want to configure H.323 softphones, How do I do that ? I am
using the Asterisk windows versio 0.60.There is no chan_h.323.so file
.Also there are no help files
hi,
i don't know if this will work or not but i've a friend working in
siemens that tell me to work with a PRI software tracer like what he
has, i still looking for a one working on linux asterisk,
using the tracer log , you can find how many digits are used :) i
don't know i wish this help
kinf
On Wed, 28 Mar 2007, John covici wrote:
This is the expected behavior -- if the second person comes in and you
have name announcements, then the first person will hear that and
should have the sense to know not to hang up. You can have everybody
hear music till a certain person comes in, if
Olivier wrote:
Hi,
Your colleague has forwarded his incoming calls to his secretary.
How do you call the feature allowing you to circumvent your colleague
call forward to make your colleague's phone ringing ?
Hi Oliver,
is this some new feature that you have invented and you need to come up
hi all,
i am using voicemailmain application in ast 1.4.2. Its not changing my
password in the change password menu. i have no idea why. my voicemail
configuration is:
25= 52,sipura
i always have to enter 52 for password even if i have changed it previously.
can anyone tell me why its not
In the extensions.conf do you have:
Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},,tT) ?
for the outgoing calls?
regards,
Staalenburg, Juan escribió:
Trixbox 1.2.3 - TDM400 FXOs - Flash (*) and # Not Working
Has anyone run into this problem. I cannot transfer or park a call (#) on
outgoing calls.
I am starting an asterisk users live conference call on Talkshoe, a
robust voIP conferencing platform I use for several podcasts. Although
I have spoken to Mark Spencer and a Digium VP about this idea, they
have nothing to do with it for the moment. They may wish to come on
board later if enough
Jay,
Just for the record, I own 3 BT102 and all three have stopped working
for various different reasons. This make me think that um... they're
not very good. Two had hardware problems, one of those was minor
(handset cord) and one will not work no matter what firmware I use.
Grandstream tried
KokMengLoh wrote:
Hi,
Does anyone know of a Video Camera that is based on SIP? There are lots
of Video Phones out there, but I can't seem to find a Video Camera.
What would you do with SIP video camera?
--
Tomislav Parcina
[EMAIL PROTECTED]
___
How to execute some system command from AMI?
--
Tomislav Parcina
[EMAIL PROTECTED]
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Stefano Corsi wrote:
is there someone who knows if I can use AOC for billing in Asterisk? I
mean: let's say I have an external SIP device that produces AOC data.
This device connects me to the telco network. Can Asterisk, if connected
via SIP with this device, collect AOC data and put it in
At 14.02 28/03/2007, you wrote:
Stefano Corsi wrote:
is there someone who knows if I can use AOC for billing in
Asterisk? I mean: let's say I have an external SIP device that
produces AOC data. This device connects me to the telco network.
Can Asterisk, if connected via SIP with this device,
I am looking for completly wireless desktop phones. Until I realized we
needed wireless i was going to use polycom soundpoint 501's. Any
suggestions on a comparable wireless phone?
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Josu Lazkano Lete wrote:
hello I want to install Asterisk just to use in my LAN, without a analog
or digital devices.
I need to install all this packages???
Asterisk 1.2.17
http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz
Zaptel 1.2.16
Yeh Jordan, my suggestion is don't.
If you read this list you'll find plenty of people complaining about
wireless functionality, the hardware/technology just isn't there yet.
Stick with wired phones and one or two wireless for particular people
for now, maybe in 12-18 month things might
Hi,
I managed to connect Asterisk 1.4.1 to my gtalk account but after
calling I hear no voice from other side (a SIP phone). Asterisk log says
nothing.
What am I missing?
TIA
Giorgio Incantalupo
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Yes, this is the output from the lsmod. I should have posted that for
clarification.
I was assuming that asterisk would have used the ztdummy module and the lsmod
command would have indicated that at least 1 program had opened the driver
interface.
I'm reading more about ztdummy now to see
Any comments on an ATA and an analog wireless? I've been doing it
that way and it works well...
Todd
On Mar 28, 2007, at 8:31 AM, Dean Collins wrote:
Yeh Jordan, my suggestion is don’t.
If you read this list you’ll find plenty of people complaining
about wireless functionality, the
Chris Nighswonger wrote:
Is anyone else on the list using Cisco 30VIP phones with the
chan_skinny driver? I have tried to catch the one of the developers on
the chat relay, but cannot seem to get anywhere.
I am trying to understand how the soft buttons are setup. They are
apparently
On Wed, 28 Mar 2007, Dean Collins wrote:
Yeh Jordan, my suggestion is don't.
If you read this list you'll find plenty of people complaining about
wireless functionality, the hardware/technology just isn't there yet.
Stick with wired phones and one or two wireless for particular people
for now,
Aastra has some new products coming that combine DECT with SIP, and look
promising. Linksys also makes an 802.11G WIFI dongle that can be mated
with their SPA-9XX series phones to untether them from your wired LAN,
and have no direct feedback on these in a commercial deployment however.
Cory
Hi Gabriele,
maybe sipp can help you: http://sipp.sourceforge.net/
Giorgio
[EMAIL PROTECTED] wrote:
Hello,
I would like to generate a peer-to-peer or a server/client
SIP traffic between two or more Openwrt access point, to
make some statistics about QoS. I tried some SIP traffic
generators
Ola Lidholm wrote:
In queue.conf (or is it called queues.conf?) you can set up a call
queue with all your phones already in it.
Which will mean that if you pass the incoming call to that queue all
phones will be ringing until one person picks it up.
At my work we have it set up like that. And
Okay, I get it. I still have a problem though. I have no way to wire 30%
of these end-points. P{hysically impossible. They do have cat3 twisted
pair to each phone. But of course they want IP. Are there any adpaters
that will give me just enough bandwidth to get it done. The computer
network is all
Meetme cant handle more than 5 users in a call?? H
http://www.voip-news.com/feature/asterisk-voip-pbx-right-choice-032707/
hmmm I'm all for commercializing a product, but this FUD from Fonality
seems to be taking it just a little too far
Regards,
Dean Collins
Cognation Pty Ltd
Does anyone know of free/cheap/open source software that will allow me to
run a test for a period of time and get an MOS score for VoIP?
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hi
After compiling and installing pwlib and openh323 ,
the asterisk, give the folloing error.
please tell me where the problem is ?
Best
Mani
*CLI -- Executing Dial(SIP/2.2.2.2-086f5ac0,
H323/[EMAIL PROTECTED]|60) in new stack
Mar 28 14:17:23 WARNING[11985]: channel.c:2576
ast_request: No
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Gordon Henderson
Sent: Wednesday, 28 March 2007 8:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] wireless desktop phones
On Wed, 28
Jordan Novak wrote:
I am looking for completly wireless desktop phones. Until I realized
we needed wireless i was going to use polycom soundpoint 501's. Any
suggestions on a comparable
I've been looking at 802.11g wireless 8 port switches. I have run into
a few hits on Google, that may
Yikes! While I will agree I think Digium needs to do a little better QA
(let's not start that war again), this kind of FUD doesn't do anything for
the community. I've had Asterisk running with meetme no problem with many
more then 5 users.
On 3/28/07, Dean Collins [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen
Sent: Tuesday, March 27, 2007 6:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Doorphone vs. Grandstream BT101
This
Klaverstyn, David C wrote:
I am using autoload and I have rebooted the server. I have tried using
different files and a different location. This is getting very
frustrating.
If the call was a SIP call then I would say that the device is using
VAD/CND (silence detection). This is the
Tomislav Parcina wrote:
How to execute some system command from AMI?
You have to login into the AMI server with proper credentials and send
commands.
I wrote an AMI test application a little while back. It gives you the
ability to login into the AMI, send commands and snoop packets
This is just a guess. I suspect the use count is counting the number
of kernel modules that are using another kernel module. Sort of a
depends on thing. i.e. zttdummy is using rtc and zaptel. zaptel is
using crc_ccitt. Since Asterisk is not a kernel module and it access
Zaptel via
Hi,
Has anybody customized* anything in Asterisk?
* Customized = Development of new features or changes the existent features.
I need a new feature in Asterisk Manager and would like to talk about this.
Thanks,
Moacir O. de Souza Junior
Belo Horizonte - Minas Gerais - Brasil
Hi Giorgio, sorry but how do this?
how i verify the server it's running, and if not runnig how i put this running.
Thanks
2007/3/28, Giorgio Incantalupo [EMAIL PROTECTED]:
Hi Carlos,
this happens to me when oppanel server is not working. Check it is running.
Giorgio
Carlos Jerónimo wrote:
Jordan Novak wrote:
Okay, I get it. I still have a problem though. I have no way to wire
30% of these end-points. P{hysically impossible. They do have cat3
twisted pair to each phone. But of course they want IP. Are there any
adpaters that will give me just enough bandwidth to get it done. The
Could it possibly be a packetization rate issue with your provider?
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote:
Many times the speed of an inbound voice call changes. It's similiar
to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible.
A speed change is the best way to
And/or periods of large jitter on your network connection.
On 3/28/07, Matt [EMAIL PROTECTED] wrote:
Could it possibly be a packetization rate issue with your provider?
On 3/27/07, Jim Duda [EMAIL PROTECTED] wrote:
Many times the speed of an inbound voice call changes. It's similiar
to
Hi,
I'm not clear on how to use Power--over-Ethernet, specifically with Polycom
phones.
What I understand, is that by buying the Polycom 501 with the 802.3af cable
bundle, I simply connect my phone, through the Polycom provided special
RJ-45 cable, into a PoE capable switch, and voilà!
Is
Responsibility for answering the door is shared by the entire office. But A) noone wants
their phone to ring, there's a door chime) and B) noone specific will accept
responsibility for answering the door. So, we need a solution that follow I'm
answering the door now, these are the buttons I
On Wed, 2007-03-28 at 11:20 -0300, Moacir O. de Souza Junior -
Personalsoft Sistemas Ltda. wrote:
Hi,
Has anybody customized* anything in Asterisk?
* Customized = Development of new features or changes the existent features.
I need a new feature in Asterisk Manager and would like to
On 3/27/07, Robert Lister [EMAIL PROTECTED] wrote:
On Tue, Mar 27, 2007 at 12:45:44PM +0200, Maxim Veksler wrote:
Hello list,
I got a couple of those wouldn't it be great questions, as following:
1. Is it possible, with asterisk to hold a central phonebook directory
of callers?, so that
A POE switch will put power on what ever line is connected to it, so if your
polycom plugs into a wall plate with cat 5 cable that runs back to a port on
the POE switch then you have power all the way to the phone.
On 3/28/07, Mike [EMAIL PROTECTED] wrote:
Hi,
I'm not clear on how to use
Aastra just released a DECT SIP solution. Supposedly they are the first to
do so but who knows. I'm not affiliated with them so it's not a plug or
anything.
http://www.aastra.com/cps/rde/xchg/SID-3D8CCB73-12C98649/04/hs.xsl/21410.htm
-Original Message-
From: Gordon Henderson
Mike wrote:
Hi,
I'm not clear on how to use Power--over-Ethernet, specifically with Polycom
phones.
What I understand, is that by buying the Polycom 501 with the 802.3af cable
bundle, I simply connect my phone, through the Polycom provided special
RJ-45 cable, into a PoE capable switch,
Dear all, I'll implement a VoIP system using Asterisk + SIP with
softphones; I need to connect LAN and VPN users (about 100-150).
What version/installation of asterisk do you recommend tyo me ??? Does
[EMAIL PROTECTED] or Trixbox match to my scenario
By the way, I use Debian Etch as OS
You don't need to change any wiring. Just be sure that the LAN wiring
terminates at a PoE LAN switch (PoE would not be passed through an
intermediate switch).
You will get an AC adapter with your phone. If the phone fails to power
up, you can plug the adapter into the thingie in the PoE
Jordan Novak wrote:
Okay, I get it. I still have a problem though. I have no way to wire 30%
of these end-points. P{hysically impossible. They do have cat3 twisted
pair to each phone. But of course they want IP. Are there any adpaters
that will give me just enough bandwidth to get it done. The
Hi Joe -
What are peoples experience with the reliability of the TDM400p. Specifically
in
the 2 FXO, 2 FXS configuration, which is the 022 (?) model.
Is this board prone to random failures?
Back to the original topic...
I have 6 of these cards installed in various asterisk installations
All,
Is there a dial plan command that can stream uncompressed audio from
another source? I see there's an MP3Player command that can stream, but
I assume that plays MP3's, which means it has to decode them. I'm
looking for something that could play .wav or .ulaw (g711) streams.
Doug.
Hi all,
I am looking for a reliable BRI (8 port) card, and I wonder which BRI
card would you guys recommend me to use?
The card will have to work in a PCI slot that is sharing IRQ with
another device...does that represent a problem (and if so, for which
cards)?
Regards,
Alex
Hi Mike -
You don't need to change any wiring. Just be sure that the LAN wiring
terminates at a PoE LAN switch (PoE would not be passed through an
intermediate switch).
One little caveat: Depending on the PoE mode, you may need to use all
four pairs of the Cat 5 cable of your network
Further disclaimer, there
is NO commercial intent behind this initiative. I only hope to bring
members of the user community together.
Someone kindly emailed privately about this. By the above disclaimer
I mean that I myself have nothing to sell in doing this not is it
meant to be a
Matt wrote:
Does anyone know of free/cheap/open source software that will allow me
to run a test for a period of time and get an MOS score for VoIP?
This one is great: http://www.testyourvoip.com
Its free and you can use it all you want. If you want to buy it to
install on your
On Mar 27, 2007, at 8:35 AM, Salvatore Giudice wrote:
As for the DSP, you are right to be concerned about the Digium cards,
but not because of the DSP. The DSP is not where you will run into
problems. Digium cards feature 2 year old circuitry and do not play
well with other devices. You have
Maxim Veksler wrote:
Thank you Rob for the detailed reply.
It solves one side of the problem (In a very cool and unexpected way I
must admit) but not the whole demand. I still would like to have a
centrally managed caller phonebook directory, available from the
phone's Directory menu. I did
Hi Steve -
Just my personal experience, but I do not find IAX to be very reliable.
Is there any particular reason you are not using SIP?
I'm curious as to your negative experiences with IAX. I generally use
it for multi-office installations, and have had good expereinces with
it. What
Steve Totaro wrote:
OK
Anyways... You could still use a Grandstream ATA and just have your
doorbell switch actually be the hook switch for the line, use the h
extension to continue ringing phones, send an SMS, jabber message or
whatever. Just set the auto dial in the ATA.
I got a
On 3/28/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Klaverstyn, David C wrote:
I am using autoload and I have rebooted the server. I have tried using
different files and a different location. This is getting very
frustrating.
If the call was a SIP call then I would say that the
No, I'm far from inventing features, yet ! ;-)
It's a feature offered by Alcatel and I wanted to find in documentation, a
way to reproduce it, just in case I'm asked to do so.
I think it's the equivalent of call screening, but from caller perspective.
Cheers
Hi Murphy,
I am developing an application for integration with Asterisk by Asterisk
Manager.
When I send a command to asterisk (Example: Action: Originate), many events
are raised. I would like to identify what events answer my command.
I'm thinking of creating a new property in the events to
Thanks for all the replies, this definitely helps me!
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller
Sent: Wednesday, March 28, 2007 12:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PoE -
I cannot seem to get any transfers to work at all. The console show I
have #1 amd #2 set up for Blind and Attended Transfer, but when I hit
these buttons on my handset nothing happens (other than I hear the dtmf
tones on the other end of the line).
roo*CLI show features
Builtin Feature
The RFP 32 access point that comes with Aastra solution reminds a product
sold by DeTeWe, a company Aastra bought months ago.
At that time, I thought it was a Kirk OEM but I've got no elements proving
it (just by looking at both products).
Cheers
___
Do you mean it c(sh)ould be included in 1.6 ? ;-)
Cheers
___
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I have some phones (and an ATA) that are shared between two users who
each have separate voicemail but they are not behaving as desired nor
expected.
Incoming calls show up on the correct lines.
Calls originating from the device are seen, at the terminating device,
as coming from the account
Alan Chandler wrote:
I cannot seem to get any transfers to work at all. The console show I
have #1 amd #2 set up for Blind and Attended Transfer, but when I hit
You need to also include the t and/or T in your dial statement.
Doug
--
Ben Franklin quote:
Those who would give up
Jordan Novak wrote:
I am looking for completly wireless desktop phones. Until I realized we
needed wireless i was going to use polycom soundpoint 501's. Any
suggestions on a comparable wireless phone?
If you enjoy being miserable and having your phones not work, by all
means, use a wi-fi
Matt,
That's possible. I've been struggling with this for a while.
I recently transitioned from cable modem service to Verizon FIOS. I didn't get
a big change in behavior ( I was hoping so ).
My VOIP provider is Teliax. My ping responses to the Teliax server are around
13/15 mS.
Can you
Jordan Novak wrote:
Okay, I get it. I still have a problem though. I have no way to wire 30%
of these end-points. P{hysically impossible. They do have cat3 twisted
pair to each phone.
If they have Cat 3 to each phone, how can it be physically impossible?
Is it *physically* impossible, or is
On Wed, 28 Mar 2007, Alan Chandler wrote:
I cannot seem to get any transfers to work at all. The console show I
have #1 amd #2 set up for Blind and Attended Transfer, but when I hit
these buttons on my handset nothing happens (other than I hear the dtmf
tones on the other end of the line).
Matt wrote:
Yikes! While I will agree I think Digium needs to do a little better QA
(let's not start that war again), this kind of FUD doesn't do anything
for the community. I've had Asterisk running with meetme no problem
with many more then 5 users.
Agreed -- they're treading on
Do you have multiple devices registering with the 10x extentions? Or is it
just the one device?
Basically, the phone is not sending the correct Caller-ID, for some reason.
Whatever caller-id the phone sends, is what will be sent.
On 3/28/07, Drew Gibson [EMAIL PROTECTED] wrote:
I have some
On Wed, Mar 28, 2007 at 01:11:05PM -0300, Alejandro Cabrera Obed wrote:
Dear all, I'll implement a VoIP system using Asterisk + SIP with
softphones; I need to connect LAN and VPN users (about 100-150).
What version/installation of asterisk do you recommend tyo me ??? Does
[EMAIL PROTECTED]
I'm also in the market for a wi-fi phone. My boss currently has a
cordless phone and wants to keep the same functionality. We have a
robust wireless network in the office and the phone will be staying
here, so roaming is not really an issue. Everybody in the office is
still going to get
Ken Williams wrote:
I couldn't find a switch, so I commented line 426 out of res_features.c and
recompiled - instant transfer now on Grandstream phones. Below is the line
for future reference.
ast_say_digits(peer, pu-parkingnum, , peer-language);
One of the many, many joys of using
Good day everyone,
Hope someone can help me with a spandsp/app_rxfax problem.
I've compiled spandsp 0.0.2pre26, and app_rxfax.c from soft-switch.org
Both went just fine, and i've checked my libtiff and libxml (along with the
devel-s) versions - they're fine.
Machine is fedora core 3, x86_64.
I'm (still) trying to get my Asterisk box talking to a Metaswitch. All I'm
getting is a heartbeat of OPTIONS messages coming from the Metaswitch
which my Asterisk box replies to. The exchange looks like:
-- SIP read from 172.b.c.d:5060:
OPTIONS sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0
I was previously having an issue with a Polycom phone and Polycom support
said that Asterisk didn't play well with Polycom firmware versions 1.6.7 and
newer due to SIP compatibility issues. I believe I heard a lot of things
were fixed\adjusted in 1.4 and was wondering if anyone has had success
On Tue, Mar 27, 2007 at 09:30:36PM -0400, Jim Duda wrote:
Lacy,
I don't have any zaptel cards installed. I do however have ztdummy
installed.
Is there some tweaks to ztdummy which I might need?
Is there a special kernel setting which ztdummy requires?
What is the output of zttest ?
Hi the list,
Think Kirk solution ;-)
www.kirktelecom.com
This is an DECT/GAP infrastructure solution, and the bases can be seen as
something like SIP/DECT gateways.
Each wireless phone is like a separate IP phone from Asterisk side.
You can use several bases and repeaters (only radio link, no
On 3/28/07, Jason Parker [EMAIL PROTECTED] wrote:
- Derek Whitten [EMAIL PROTECTED] wrote:
if i remember right, most of the buttons on those and the 12SP+ phones
don't really work
because there isn't a button template in *
There is a button template, the problem is that most of the
How do I clear a global variable for good? I have a situation of
needing to use global variables to aide in channel communication, but
will be changing the name within a defined scope.
Additional Background...
I want to get a variable from a channel (child) that is created by
another channel
Start with a codec check (sounds like the CNG tone frequencies are out of
spec)...
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Wulter
Sent: Wednesday, March 28, 2007 4:58 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
I would be interested in specifics as I have yet to hear any real
issues, a lot of people had some bad taste after 2.0.0, as is to be
expected for a first release.
I've used 2.0.2, 2.0.3, and now 2.1.0 with Asterisk 1.2 for months
without issues.
From: [EMAIL
On Wed, 2007-03-28 at 15:55 -0300, Moacir O. de Souza Junior -
Personalsoft Sistemas Ltda. wrote:
Hi Murphy,
I am developing an application for integration with Asterisk by Asterisk
Manager.
When I send a command to asterisk (Example: Action: Originate), many events
are raised. I would
Matt,
I am running Polycom 2.1 on both 1.4 and 1.2 svn releases without any
problems. What kind of issues did you experience?
On 3/28/07, Mike Hammett [EMAIL PROTECTED] wrote:
I was previously having an issue with a Polycom phone and Polycom support
said that Asterisk didn't play well with
Just be careful with any multi vendor GAP solution (GAP is Generic
Access Profile - which means you are supposed to be able to take a
handset from any vendor and match it with a base station from any
vendor)
Basically it's like any standardsure you get basic functionality but
you'll often
Dean Collins wrote on 3/28/07 9:27 AM:
Meetme cant handle more than 5 users in a call?? H
Heh, that's a laugh. We regularly get 40 or more callers in a
conference room in MeetMe with no problems. In fact, the call quality
is better than some of those 800# conference services we used to use
Evnin'
As I didn't find any answer I'll try to rephrase the problem (o;
Any idea why the latest asterisk-addons-1.4 write wrong uniqueid
into mysql database?
Asterisk-1.4.2 creates call record files with the uniqueid
prepended:
1175107269-SIP-999-0876c000.wav
But into mysql
Oh poo. No one seems to know. :(
Doug Garstang wrote:
All,
Is there a dial plan command that can stream uncompressed audio from
another source? I see there's an MP3Player command that can stream,
but I assume that plays MP3's, which means it has to decode them. I'm
looking for something
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