Olle E. Johansson wrote:
Chris A. Icide wrote:
Note however, that in the unsolicited NOTIFY that Asterisk sends for
MWI, it includes a ;tag= as part of the NOTIFY. This will break some
devices as they will not accept the NOTIFY because the tag doesn't match
any transaction that
The comparison to LiveVOIP was in relation to ShellTel, not to
VoipSupply.
-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 19, 2005 7:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: So you all
That is correct -- the comparison was to SmellTel, not to VoipSupply.
I have not dealt with VoipSupply personally, but from the professional
public responses to unprofessional public attacks, I don't doubt that
your business future is bright.
Much brighter than some of your customers are...
Chris A. Icide wrote:
Olle E. Johansson wrote:
Chris A. Icide wrote:
Note however, that in the unsolicited NOTIFY that Asterisk sends for
MWI, it includes a ;tag= as part of the NOTIFY. This will break some
devices as they will not accept the NOTIFY because the tag doesn't match
any
It would be a good feature for the blind. Lets say they are away from
the home/office and don't have access to their computer. They could
check important email via a TTS/IMAP
interface.
Dean Collins wrote:
Ah but of course if he had a treo then he would be checking his email
via
I have found my PAP2-NA's to be picky about their DHCP server. My PAP2-NA
appears to lockup if it is set for DHCP and the server is my Netgear RP614
Websafe router. The fix for this is to unplug the ethernet from the unit,
plug in an analog handset, power it on, using the handset perform a
Hi all,
Some simple questions about codecs:
What codec does the Zap channel use by default?
Can this default be changed, and to what? (g729 too?)
What codec does meetme use? (I think this is ulaw, but asking to be sure)
Can you use another codec, or does everything have to be transcoded to
Hi,
have you tried to open asterisk consolle and type set verbose 9??
Giorgio Incantalupo
Leo Burd wrote:
Hello everyone,
I'm trying to send debugging messages to the console. However,
although my system seems to be working fine, it does not seem to be
printing the NoOp messages on the
Hello list,
Did anyone already get the T410P card running in an
HP-Compaq DL380 G4 server? If yes, how?
I'm using Fedora Core 3 with 2.6.11-1.35_FC3smp Kernel package.
Thanks in advance,
Roland
___
Asterisk-Users mailing list
Hello All,
I have an iaxy(new version), and while it does the job well, there is
one thing I am looking for. I want to be able to specify a dns name on
the config, not an ip. This does not seem to work if I try to set it as
such. Has anyone come up with a workaround or solution to this?
I want
Some time in the future, on Wed, Jul 20, 2005 at 06:03:46PM +0800, chris wrote:
hi,
im installing latest asterisk from cvs on solaris 9. but when i run make i
got this error
/bin/sh: build_tools/make_version_h: cannot execute
make: *** [include/asterisk/version.h] Error 1
what i did
Sorry but can u remind me which problem?? i've answered many people before with diffrent problems what problem u r having?
On Tue, 2005-07-19 at 15:20 -0400, Jerry Rasmussen wrote:
Can you help me remember the details. Its been a while since I have touched the system.
What is your dmesg output when you fire up the card.
There were some problems with TE410P and the intel chipset used in the
DL380 G4's.
You need firmware at least 'TE410P version c01a010b'
Contact Digium and RMA if you have older firmware (basically the symptom
will be everything
is ok,
pbo 808 wrote:
I've done quite a bit of googling and haven't found a solution to my problem.
I've got the Digium dev kit (wctdm11b) set up and working. I've
compiled spandsp and can receieve faxes from eFax (www.efax.com) but
the pages are blank. The page count is correct, in that if I fax
hi Tzafrir,
yes i have all the files
bash-2.05# pwd
/usr/cvsroot/asterisk/build_tools
bash-2.05# ls
CVS make_defaults_h mkdep
make_build_h make_version_h vercomp.c
bash-2.05#
i alos executed chmod -R o-w /usr/cvsroot/asterisk/build_tools/
bash-2.05# ls -l
total 30
On Tue, 2005-07-19 at 07:12 -0400, Chris Mason (Lists) wrote:
Madhawa Jayanath wrote:
o Bernie,
1) best results www.nufone.net
2) low cost www.voipjet.com
Anyone able to find NuFone's rates? I have been looking for them on
their site. I need international rates and UK Mobile.
As
Hi list
I plan to implement a prepaid solution where the system needs to check for
remaining credit periodically during a call. The reason for this is that this
is a system where the credit pool can be used simultaneously by more people,
and not only for calling.
I have a problem figuring how
On Tue, 2005-07-19 at 12:42 -0700, Derek Whitten wrote:
rofl..
nufone sends you configuration information via email after you sign up
for an account..
On Tue, 2005-07-19 at 11:16, Andrew Kohlsmith wrote:
Nufone seems to have always been a DIY type of VOIP provider. Their new
members
Hi,
I'm experimenting attended calls tranfers and I'm a little bit
confused.
In usual pbx's normaly there is no difference between an attended call
transfer and a blind one:
you just hit transfer then dial the extension you want the call to be
transfered.
If you stay on the phone you can talk
Hello,
I have a client that has a fairly small installation (20 SIP
Phones) that is running Stable. Asterisk appears to be consuming large
quantities of memory, and growing uncontrollably to the point where after
about 6 weeks the box starts to swap itself to death. I've been keeping my
Hi,
we are trying to install Junghann's quadBRI into Dell PowerEdge 2800
system without success.
I don't know if the issue can be that Junghann's card fits 32-bit slot
and Dell PE 2800 has
only 3 PCI-X 64-bit slots. Can this be an issue?
We get CRC errors for HDLC frame when the card is
I've recently upgraded to Asterisk 1.0.9 on a system which gets frequent use,
mainly for MeetMe calls. After running for a while (a few days), the channel
list gradually accumulates a bunch of these Zap/pseudo thingies, which just sit
in the channel list forever. This accumulation is
Alessio Focardi wrote:
Hi,
I'm experimenting attended calls tranfers and I'm a little bit
confused.
SNIP
I honestly think that transfers is one thing that Asterisk should
improve a LOT to be able to stand up to even the most cheapo taiwanese
no-name PBXs, which support attended transfers
Hi, all
I have Polycom SP300 phones.
Calls between those are ok and quality is great.
Then I have IAX2 soft phones (FireFly). Calls between those are OK too.
But when I have call b/w Polycom (SIP) and IAX, I have really bad echo at
Polycom phone side. IAX phone side is OK.
Any ideas?
I had the same problems with a 4 port junghanns and a 4 por wcfxs
I took the junghanns out and added it into a new box and all was ok
So ether it was because the 2 cards was in together or it was the
motherboard?
U using the latest driver and asterisk?
On Wed, 2005-07-20 at 11:31 +0200, David
Hello Michael,
Wednesday, July 20, 2005, 11:54:40 AM, you wrote:
MP Alessio Focardi wrote:
Hi,
I'm experimenting attended calls tranfers and I'm a little bit
confused.
MP SNIP
MP I honestly think that transfers is one thing that Asterisk should
MP improve a LOT to be able to stand up to
Dear
All,
I want to get a
clear picture on howreal time users/peers acts. in the case were 2
*servers acting for failover. *1goes offline *2 takeover
tocompletes the SIP session started with *1 without the need to resets the
call.
Is this attitude is
achieved by using real-time Asterisk
Hi All,
I am looking for a GSM VoIP gateway for use with
Asterisk. I have come across VoiceBlue by 2N but it's
price is beyond my reach. Are there any other
alternatives out there?
I've scanned across the mail achieves for an answer to
this without much success, if the question has already
been
Thank you very much for your support. Now I can pick up the distant
ringing phone with my Cisco 7960 IP Phones(with SIP image). But in
case of 7910's with SCCP image I cannot do that. Does SCCP support
pickup group?
C F, I am actually confiused by searching Follow Me. I have found that
Follow me
Hi All, I am just looking at using Asterisk now and the first thing I need to do is via pass two external numbers to asterisk and call out connecting the calls togther. These will be through our physical PBX connected to the asterisk server. We are essentially trying to connect two external
I'm sure its been brought up previously on the list but I personally don't
think that TTS is very practical due mostly to all the crap that gets
stuffed into emails these days. How do you handle:
RTF
HTML
Disclaimers
Signatures (inc ascii art sigs)
Virus scanner tags
The 1ste pc I tried it on was on a expensive intel board and the second
one that worked was on some cheap name board
Ill say incompatibility ?
Yes, I do use latest bri-stuff package (asterisk 1.0.9 incl)
Any ideas?
-
David Hajek
IT/IS Manager
Systinet Corporation
Phone: +420 2 7201 9526
Hi to all,
I have been using Wildcard TDM400P with four fxs modules and it was
working fine now I have added another Wildcard TDM400P with four fxs
modules . So there are total 8 ports for 8 hard phones.
I have modified following configurations
In /etc/zaptel.conf
loadzone=us
defaultzone=us
Exactly.
Junghanns works for me on old PIII system, but it does not work in
newest Dell PE.
-
David Hajek
IT/IS Manager
Systinet Corporation
Phone: +420 2 7201 9526
Cell: +420 604 352 968
[EMAIL PROTECTED]
http://www.systinet.com
Altus Snyman wrote:
The 1ste pc I tried it on was on a
On Wed, Jul 20, 2005 at 04:22:01AM -0700, Mazhar Hussain wrote:
Hi to all,
I have been using Wildcard TDM400P with four fxs modules and it was
working fine now I have added another Wildcard TDM400P with four fxs
modules . So there are total 8 ports for 8 hard phones.
I have modified
HI ALL;
Is there any free Ipivr application for Asterisk? I
heard about Simeda but its site is not accessable
Appreciate any help
Mohammad
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On Wed, Jul 20, 2005 at 07:01:48PM +0800, Craig Guy wrote:
I'm sure its been brought up previously on the list but I personally don't
think that TTS is very practical due mostly to all the crap that gets
stuffed into emails these days.
start with the simple stuff
How do you handle:
Allan Kamau schrieb:
...
I am looking for a GSM VoIP gateway for use with
Hi,
do you think of something to interconnect
to GSM carriers via cable (GSM-A) or do you
think about using a GSM-modem with all its
limitations?
For the first option I could forward your email address
to someone
Hi,
I was wondering if somebody could help me with getting Asterisk to SEND a fax.
Here is what I want to do:
We have three offices and people always have trouble getting to it, so I want to have a simple way to people to call in and ask for a map for office A, B or C.
The way it would work is
For the second option, it might be interesting for you,
that we are currently also working on asterisk support
for a GSM-modem.
If that can be used to build cost effective, quality GSM gateways which
can be used for call termination, I'm interested.
- What GSM modules do you plan to use?
-
What revision of card is the new one? It sounds like you have one of the
new Rev I cards and you aren't running either 1.0.9 or CVS HEAD. Either of
these will solve your problem if I am correct.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi,
I am unable to get a dialtone from iaxy (the old model). When dial a
mailbox, I can see the mailbox app reacting.
iaxy gets registered. I can make call and the remote phone can hear me. No
sound for iaxy user.
./iaxyprov 192.168.1.134 provinfo
01:
05:
11 d9
0d:
Some simple questions about codecs:
What codec does the Zap channel use by default?
None/all. Look in /usr/src/asterisk/configs/zapata.conf.sample and
you won't find any reference to codecs. Think of the zap channels
sort of as a layer-two protocol interacting with external devices
such as
I just wanted to say that our company is more then happy with the
service we have received from voipsupply and have now purchased over
25 ATA units from them to start our VoIP business.. and plan to
purchase several hundred more in the coming weeks never a
problem.. always next day delivery
On Wed, 2005-07-20 at 14:42 +0300, Tzafrir Cohen wrote:
On Wed, Jul 20, 2005 at 07:01:48PM +0800, Craig Guy wrote:
How do you handle:
RTF
Not very common
Isn't this easily converted to text?? I thought the format for this was
pretty simple, but I could be wrong...
Disclaimers
Hi,
This is exactly the set up I was trying and could not get it to work (also
w/ efax and the dev kit). You got slightly farther than I did -- for me,
the detection worked and the macro launched for the extension, however the
tiff file never got created.
I did some additional searching in the
On Wed, 2005-07-20 at 12:26 +0200, Alessio Focardi wrote:
Wednesday, July 20, 2005, 11:54:40 AM, you wrote:
MP Alessio Focardi wrote:
I'm experimenting attended calls tranfers and I'm a little bit
confused.
MP I honestly think that transfers is one thing that Asterisk should
MP improve a
hi,
I know this topic has been covered several times and I read everthing
I found on the subject but I restarted one of my cisco phones and it
suddenly gave the famous message :protocol application invalid
how can I use ethereal to see what's going on ? anybody could give
me the command
On Wed, 2005-07-20 at 10:49 +0200, Eivind Trondsen wrote:
Hi list
I plan to implement a prepaid solution where the system needs to check for
remaining credit periodically during a call. The reason for this is that this
is a system where the credit pool can be used simultaneously by more
Thanks Roger, I find the second option more
interesting, let me know once you've managed to
provide asterisk support for the GSM modem.
Allan.
--- Roger Schreiter [EMAIL PROTECTED] wrote:
Allan Kamau schrieb:
...
I am looking for a GSM VoIP gateway for use with
Hi,
do you think
Amr Shaheen wrote:
Dear All,
I want to get a clear picture on how real time users/peers acts. in the
case were 2 * servers acting for failover. *1 goes offline *2 takeover
to completes the SIP session started with *1 without the need to resets
the call.
Is this attitude is achieved by
[EMAIL PROTECTED] wrote:
I have an iaxy(new version), and while it does the job well, there is
one thing I am looking for. I want to be able to specify a dns name on
the config, not an ip. This does not seem to work if I try to set it as
such. Has anyone come up with a workaround or solution
chris wrote:
bash-2.05# ls -l
total 30
drwxr-xr-x 2 root other512 Jul 19 12:33 CVS
-rw-r--r-- 1 root root 405 Jul 19 12:33 make_build_h
-rw-r--r-- 1 root root 983 Jul 19 12:33 make_defaults_h
-rwxrwxr-x 1 root root 181 Jul 19 12:33
Kevin P. Fleming wrote:
[EMAIL PROTECTED] wrote:
I have an iaxy(new version), and while it does the job well, there is
one thing I am looking for. I want to be able to specify a dns name on
the config, not an ip. This does not seem to work if I try to set it as
such. Has anyone come up with
The iaxy doesn't support dns. Its a very expensive little box with very
little features, unfortunately.
Tim
[EMAIL PROTECTED] wrote:
Hello All,
I have an iaxy(new version), and while it does the job well, there is
one thing I am looking for. I want to be able to specify a dns name on
the
That may be true, but for whatever reason I cannot get through to the
phone number that they advertise on the front page. I setup an account
and had weird stuff happening throughout the day, no way to get ahold of
them, and no response to email yet, even though they claim 24/7 support.
I am
David Hajek wrote:
We get CRC errors for HDLC frame when the card is initialized. Any
idea what can be wrong?
After loading the driver we got CRC errors like this:
Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card
1 (cardID 0) S/T port 1
What are your settings in
Hello
I see it is possible to buy Flash Disks up to 4GB
now. Has anyone any experience of building an Asterisk system with a flash
disk as the only storage device? Any brands you recommend? Is 2 or
4GB enough for an Asterisk installation? Typically how many MB is required
for voicemail
Hello all,
I've got a question regarding automatically logging in an agent on a
predefined extension. I want to provide agents with a webinterface they can
use to login in, now you might wonder why they can't use the phone - don't
ask - customers eh :-)
So what I basically need is a way to
Can anyone shed any light on an issue with agent penalties?
I have 2 queues set up with agents working both queues, but where agent
1 should have a penalty for queue 2 and agent 2 should have a penalty
for queue 1. When a call is sent to either queue, it rings agents with
and without penalties
Hi
all,
I'm currently
gearing up for a possible PBX replacement project using Asterisk, and I'm just
breaching the iceberg of information that's available.I typically
like to have something thick with pages in front of me. Mahler's book was
the first one to come up and it seems like a
Greetings,
We have a solid state Asterisk server provided by The VOIP Connection
that uses a 512MB Compact Flash device as the primary boot device as
well as the main storage device and it works fine for us. The system
isn't the fastest booting, but the technology is absolutely sound. Make
Hi all.
I am trying following scenerio for call park pickup.
voice is flowing established between B C, after call-pickup (
instead of A B ).
can anyone please clarify why it is happening like this, ( or ) do
i need some more configuration for parkpickup ?
A
B C
Talking
Goto their website and buy it. www.signate.com I know paul he's a good guy.
Has a new book coming out soon.
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
From:
Personally I wouldn't bother with the Mahler
book. I bought it in the hope that it might be the panacea I was looking
for. It wasn't. If you read it you will recognise a lot of the
standard text you will see on Digium or other web sites.
If I had time I would write the book
myself.
I did
David Stude wrote:
Hi all,
I'm currently gearing up for a possible PBX replacement project using
Asterisk, and I'm just breaching the iceberg of information that's
available. I typically like to have something thick with pages in
front of me. Mahler's book was the first one to come up
David Stude wrote:
I'm currently gearing up for a possible PBX replacement project using
Asterisk, and I'm just breaching the iceberg of information that's
available. I typically like to have something thick with pages in
front of me. Mahler's book was the first one to come up and it seems
So what I basically need is a way to log in an agent using
AgentCallbackLogin
at an extension without them having to answer / pickup a phone to do so. I
looked at the Manager API but did not find any command related to agent
logins.
Yes even with latest CVS there are no Manager Actions for
Guys I just read on the wiki:
2005-07-19 - long awaited extension lights (hint priority) and call pickup
on various phones work with newly released asterisk patch digium bugtracker
- feel free to test and report findings to the bugtracker to have this
commited to cvs.
How does this work? And
You can try at:
http://www.voip-info.org/wiki-Asterisk
regards.
- Original Message -
From:
David Stude
To: asterisk-users@lists.digium.com
Sent: Wednesday, July 20, 2005 3:56
PM
Subject: [Asterisk-Users] Mahler's Book -
New Project
Hi
all,
I'm
Hi all.
I am trying following scenerio for call park pickup.
voice is flowing established between B C, after call-pickup (
instead of A B ).
can anyone please clarify why it is happening like this, ( or ) do i
need some more configuration for parkpickup ?
A
My zaptel.conf:
[channels]
switchtype = euroisdn
; BRI CARD
nationalprefix = 0
internationalprefix = 00
signalling = bri_cpe
pridialplan = local
prilocaldialplan = local
echocancel = yes
rxgain=-2
txgain=4
echotraining=yes
context=from-isdn
usecallerid=yes
hidecallerid=no
group = 1
channel =
Anton Krall wrote:
How does this work? And will it work only on certain phones or can it work
with the gxp2000?
I think for the GXP-2000 the firmware also needs support for this. (and
it's planned for a future version)
Cheers,
Kristof.
___
I need to place a call using a pin code. To access an external line,
the host PBX (a Ericsson MD-110) will require that I dial
*72*pincode#phone_number to complete any (trunk) call.
When I send the number, my Sipura 3000 will reject the call with
Forbidden - wrong password on authentication
On Wednesday 20 July 2005 15:56, David Stude wrote:
Hi all,
I'm currently gearing up for a possible PBX replacement project using
Asterisk, and I'm just breaching the iceberg of information that's
available. I typically like to have something thick with pages in front
of me. Mahler's book
it's fixed
russell at lists.digium.com russell at lists.digium.com
Tue Jul 19 11:23:06 CDT 2005
a.. Previous message: [Asterisk-cvs] asterisk utils.c,1.58,1.59
b.. Next message: [Asterisk-cvs] asterisk utils.c,1.59,1.60
c.. Messages sorted by: [ date ] [ thread ] [ subject ] [ author ]
David Hajek wrote:
[channels]
switchtype = euroisdn
; BRI CARD
nationalprefix = 0
internationalprefix = 00
signalling = bri_cpe
Ouch, we are talking about zapata.conf I guess :-)
But, that all seems okay.. Have you tried signalling = bri_cpe_ptmp
instead of bri_cpe?
Do you use this card in
Hi,
alhtough i googled for details concerning ASTCC i did not found an aswer to
the following:
Should i put in my extensions.conf the configuration of the astcc? I ask
this because as i see it, in the end of the extensions.conf there is an
include statement :
#include
Kristian Kielhofner, the lead developer of the AstLinux project, will
be speaking at ClueCon. His latest AstLinux Version 0.2.6 is a
complete Asterisk distribution built to run from Compact Flash and
uses less than 32mb.
Thanks,
Brian
___
Hi i installed ceptral and i want to test it with asterisk can u plz tell me if i was wrong here ??
exten = 2,1,Answer
exten = 2,2,system(/opt/swift/bin/swift hello world)
exten= 2,3,Hangup()
Mahmoud Badran
ATSI
Tel: +20 2 607 8917
Hello all,
Kevin P Fleming once said that a patch will be released very soon to
send Remote-Party-ID header from Asterisk. and this was said probably in
Feburary.
is that patch released yet or not ? if some please comment, I will
really appriciate
Regards,
--
Atif
On Wed, Jul 20, 2005 at 04:38:07PM +0200, David Hajek wrote:
My zaptel.conf:
/etc/asterisk/zapata.conf, you mean?
And I'd like to use this opportunity to introduce a newer version of
genzaptelconf. It should basically identify zaphfc and qozap as well
as E1 and T1 (hopefully).
Does anyone know how to get rid of these hung channels?
I am getting this when I do a:
show sip channels
209.82.xxx.xxx 0071495217 2591218534@ 00103/1 unknow(d)
209.82.xxx.xxx 0041590104 0690231739@ 00103/1 unknow(d)
209.82.xxx.xxx 0070259259 3265102826@ 00103/1 unknow(d)
Title: Message
It
won't pay to try to do anything too quickly with Asterisk. You will want
to take your time and learn the ropes, like somebody else suggested, setting a
system up with a telephony card andsome inexpensive phones or ATAs will
show you what you are in for.
I
bought Mr.
I'm currently gearing up for a possible PBX replacement project using
Asterisk, and I'm just breaching the iceberg of information that's
available. I typically like to have something thick with pages in
front of me. Mahler's book was the first one to come up and it seems
like a good place to
For the fella who wanted MOH music.
Royalty free stuff can be found here.. The Acoustic Guitar
is a nice collection
http://www.freeplaymusic.com/
Cheers,
W
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi everybody,
I'm new to this matter and I spent three days in trying to connect one SIP
Softphone to an Asterisk Box. I always get error 401 or 403...
I don't understand very well settings in Softphone program: con anybody show
me how to set up a minimal running system with no public lines or
I am having a problem with my CDR web page (AMP). There is a banner on
the page saying YOu MUST ACCESS THE CDR THROUGHT THE ASTERISK
MANAGEMENT PORTAL! and it will not show any calls just No calls in
your selection. I have checked the database and calls are being
recorded in the database.
I
All,
I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones.
All seems well other than the phones have to be reset up to 5 times per day.
It is like they lose thier ip connection or maybe thier SIP connection. Has
anyone else experienced this issue? I have the phones
Hi,
I spot weird behaviour of latest Firefly 3rd party on my laptop. Sometimes
it comes to state that it won't start (hangs on Initializing ) and it
again works after system restart... Didn't yet figured out how to recreate
it.
Any similar experience ?
Also - how can I force Firefly to
On Wed, 2005-07-20 at 08:26 -0400, Ousmane Doukara wrote:
Hi,
I am unable to get a dialtone from iaxy (the old model). When dial a
mailbox, I can see the mailbox app reacting.
iaxy gets registered. I can make call and the remote phone can hear me. No
sound for iaxy user.
./iaxyprov
Angus Comber wrote:
Hello
I see it is possible to buy Flash Disks up to 4GB now. Has anyone
any experience of building an Asterisk system with a flash disk as
the only storage device? Any brands you recommend? Is 2 or 4GB
enough for an Asterisk installation? Typically how many MB is
Hello
I am planning to build a small PBX using
TDM22B.
We have a Siemens Hipath 3750 in operation
already.
When I manage to complete my PBX using TDM22B
I would ofcourse like to be able to connect my Asterisk PBX
with the Siemens Hipath 3750 PBX.
Any
Brian West wrote:
Kristian Kielhofner, the lead developer of the AstLinux project, will
be speaking at ClueCon. His latest AstLinux Version 0.2.6 is a complete
Kristian will also be speaking at Astricon 2005 in California
http://www.astricon.net/2005/
/O
On 11:31, Wed 20 Jul 05, David Hajek wrote:
Hi,
we are trying to install Junghann's quadBRI into Dell PowerEdge 2800
system without success.
I don't know if the issue can be that Junghann's card fits 32-bit slot
and Dell PE 2800 has
only 3 PCI-X 64-bit slots. Can this be an issue?
We
Title: Message
Thanks for your help, everyone. I think I've
decided to go without the book- I agree that everything I need seems
pretty much out there and I'm not very patient with poorly-written printed
material, anyway.
Thanks and looking forward to taking part
here.
-Dave
Hi everyone,
i'm announcing YAACID: Yet Another Asterisk Caller ID V0.9
it's the first public release. we've been using it in our office for a
couple of months now.
YAACID is a native Windows (.NET) program that sits in the
notification area and logs into the manager interface. it waits for a
Cavanna, Richard wrote:
I am having a problem with my CDR web page (AMP). There is a banner on
the page saying YOu MUST ACCESS THE CDR THROUGHT THE ASTERISK
MANAGEMENT PORTAL! and it will not show any calls just No calls in
your selection. I have checked the database and calls are being
I've been using the extension lights on my polycoms before that patch,
so I'm not sure what it fixed, but I've only seen the lights work on
Polycoms and Snoms. Try using the hint priority and see if it works
for your gxp2000, be sure to post your results!
--
Tom Hayden
Astoria Telecom, LLC
On 16:06, Wed 20 Jul 05, [EMAIL PROTECTED] wrote:
All,
I have AAH 1.0 installed using Digium TDM04B and Grandstream GXP2000 phones.
All seems well other than the phones have to be reset up to 5 times per day.
It is like they lose thier ip connection or maybe thier SIP connection. Has
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