On Wed, 2 Mar 2005, Turgut Abacioglu wrote:
Hello
I downloaded Astwind and get working the network (means can access to
Internet through MS Windows). DEbian and Asterisk files are updated from
Internet. But When I make install in Zaptel (it was my first make) I got
many errors. Acoording
On Wed, 2 Mar 2005, Nik Martin wrote:
Due to the unfortunate nature of Wikis, the section on voip-info.org
that deals with dual asterisk servers is full of pretty bad and
outdataed examples.
What I'm trying to do is distribute small asterisk boxes to remote
offices that have SIP clients
On Sat, 12 Feb 2005, Steve Underwood wrote:
[ DELETED ]
How do you control acoustic spill within a phone through the use of
directional microphones? Adjusting gains mitigates the issue a bit, but
is hardly a solution. These are just bodges, not solutions.
And just to add my two cents to the
On Sat, 12 Feb 2005, Robert Hajime Lanning wrote:
[ DELETED ]
ok, I did miss that. Then again, the grandstream does have a
speaker phone. I guess the problem is that I don't know of a
SIP hardphone that doesn't have a speaker phone.
Polycom IP300 has a Listen Only speakerphone. No
On Thu, 3 Feb 2005, Brian Dingman wrote:
I took them up on their offer for a refund. IMHO they shouldn't offer
* service at all. Even outgoing calls aren't handled properly. Lots of
making progress - no answer results.
Others have suggested iax.cc. However, they haven't repsonded to my
On Sat, 29 Jan 2005, Andrew Kohlsmith wrote:
On January 29, 2005 11:29 pm, Brian Dingman wrote:
This is driving me crazy. I have resorted to using the m option in the
Dial command just so folks don't hang up. I can't believe nobody else
is having this issue.
Simple test: try it with
On Sun, 30 Jan 2005, Remco Barende wrote:
You'll be better off buying the TDM400 from Digium.
I just ditched all my X100P clones because of huge problems with echoes.
Even worse, the interrupt problems with the clone cards caused my box to
crash under heavy I/O.
Want to buy the X100P
On Sun, 30 Jan 2005, Paul Tyreman wrote:
Hi,
This might not be a very popular question, but I was just wondering if
anyone have ever tried to run Asterisk on a Windows computer using Microsoft
Virtual Server
(http://www.microsoft.com/windowsserversystem/virtualserver/default.mspx).
On Sun, 30 Jan 2005, Paul Tyreman wrote:
http://www.digium.com/index.php?menu=astwind
I think this may be worth a look, I'm downloading it as I type this
e-mail...
I didn't know Asterisk had the possibility of being run on a windows machine
and while it's not as stable as a Linux
Hello,
I'm looking at building a couple new PRI Gateway boxes using
TE405P cards, and was wondering if anyone has had any experiences (good or
bad) with the Intel SE7210TP1_E motherboards from Intel. General Technics
builds some really nice (and cost effective) 1U servers based on the
Hello,
I've got a mixture of SPIP 300 and 500 phones in production for
various clients. I've got the XML settings configured for local
conferencing, but I'm not seeing the expected behavior from the phone when
I attempt to conference two calls together. According to the manual, while
Hello,
I have a mixture of Polycom SP IP 500 and 300 phones. I have been
reading through the administration manual to try and solve this problem,
but I do not seem to be able to find the answers to my question. I figured
I would ask here and see if anyone has some suggestions.
The
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein
Sent: Wednesday, January 12, 2005 12:54 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom IP 500 Dial Issues
Hello,
I have a mixture of Polycom SP IP 500 and 300 phones. I have been
reading
On Wed, 12 Jan 2005, Paul Rodan wrote:
Yeah, it's a way for numbers to get sent faster, so you don't have to wait
for the 3 second timeout before it gets transmitted to Asterisk. It's
similar to the dial-plan in the Sipura devices.
I don't know where it's mentioned in their documentation,
On Wed, 12 Jan 2005, Andrei (MPI) wrote:
Greg Boehnlein wrote:
Hello,
I have a mixture of Polycom SP IP 500 and 300 phones. I have been
reading through the administration manual to try and solve this problem,
but I do not seem to be able to find the answers to my question. I figured
On Wed, 12 Jan 2005, Kai-Uwe Jensen wrote:
On Wed, 12 Jan 2005, Paul Rodan wrote:
Yeah, it's a way for numbers to get sent faster, so you don't have to
wait
for the 3 second timeout before it gets transmitted to Asterisk. It's
similar to the dial-plan in the Sipura devices.
I
On Wed, 15 Dec 2004, Matt Klein wrote:
3) good luck getting the firmware source
is the firmware source freely available, -- I've been asked by others.
All the other (excellent, thought provoking) conversation aside, Jake
Messenger from Portmasters.com has been granted a license by Lucent
On Wed, 15 Dec 2004, Matt Klein wrote:
W/ Portmaster to Ether
Portmaster
Qty 2 T1 $400
Portmaster
Qty 2 T1 $400
Carrier Access
Qty 4
On Wed, 15 Dec 2004, Matt Klein wrote:
who said anything about a computer? :) computer, $$extra on both.
may be less on the pm3 side due to resource needs.
In the scenario I envision this being used in, there is no computer. The
PM3 runs (On it's x86 w/ 4 or 16 megs of ram) a stripped
On Mon, 13 Dec 2004, Damon Estep wrote:
http://www.millenigence.com/articles/asterisk-non-technical-review.pdf
A word of warning..
That document is a good start, but you best not put it in front of a
decision maker. There are a bunch of grammatical and typographical errors
that need to be
On Mon, 13 Dec 2004, Christopher Dobbs wrote:
Had not thought of that.
We are mostly interested in providing POTS lines inside of a company.
Think of it as a a massive CPE device running on the VPN of a
corporation with offices in many towns.
who knows?
We saw all of the traffic
On Fri, 10 Dec 2004, JR Richardson wrote:
Hi Guys,
The article They've Got Your number in the Dec 2004 issue of WIRED
magazine mentions Asterisk PBX (on p.100). The article is about phone
phreaks hijacking cell phones with Bluetooth technology along with spoofing
CID to pull some
On Fri, 10 Dec 2004, nik martin wrote:
news.gmane.org wrote:
Allied Telesyn VoIP Access Device
http://www.alliedtelesyn.co.uk/site/files/documents/datasheet/VP624FXS_euro.pdf
This is a 24-port FXS 1u device, conveniently presented as a single
RJ-21 TELCO connector.
yeah,
On Thu, 9 Dec 2004, Jorge Mendoza wrote:
Andrei,
I'm interested too. Any chance to put the archive in a ftp site?.
Jorge Mendoza
I am also interested in getting the 1.3.4 firmware. It annoys me that I
can't just get it from Polycom's website, and forces me to rethink
deploying their
On Tue, 7 Dec 2004, Brian West wrote:
If u used
allow=G723.1 , now u have to use : allow=g723 * ( NO CAPS ) !!!
allow=G729 , now u have to use : allow=g729 * ( NO CAPS ) !!!
allow=iLBC , now u have to use : allow=ilbc * ( NO CAPS ) !!!
Please , verify all configurations files (
On Wed, 1 Dec 2004, Kanuri, Seshu (Company IT) wrote:
Tell me which one can get me access to the LinkSys Linux using SSH?
Does Satori has this feature? I am not so concerned with Voice Shaping
and QOS at this time, but more interested in converting this into a
Linux box that is
On Thu, 2 Dec 2004, Eric Rees wrote:
Where only talking about 100 extensions. That is a lot to hard code by
hand.
Just use app_queue and define a list of members as the SIP extensions. It
is a lot easier to maintain the queues.conf file than to worry about
adding 100 extensions into your
On Mon, 6 Dec 2004, Steve Kennedy wrote:
Well, then roll your own and stop whining about it. Quite frankly calling
it hackware shows that you have no concept of how much work has gone
into the Sveasoft firmware, nor do you grasp the concept that Linksys is
incorporating many of the
On Mon, 6 Dec 2004, Kanuri, Seshu (Company IT) wrote:
Greg Wrote
/SNIP/
Everyone wins from this, and Sveasoft has a revenue stream that allows
them to keep focused development on improving the firmware. I have over
60 of the WRT54GS units in production and I run Sveasoft firmware on
every
On Wed, 1 Dec 2004, Samudra E. Haque wrote:
I would like to build my newest server based upon Fedora Core 3, and
load up asterisk. I was all set to do so.. but then I read in Asterisk
Users Digest, Vol 4, Issue 404:
I think you would be insane to run your production servers on Fedora Core
My copy of VoIP Business Weekly came today and the main article on the
front page is titled: Penguin to Assault PBX Market. Their premise is
that Linux is driving the adoptation of Open voice standards in the
market, and as such is poised to disrupt the dominance of market leaders.
Third
On Fri, 26 Nov 2004, Rich Adamson wrote:
[SNIP]
What if you have a single port T1/E1 card from digium? No decision to
be made really; you're going to sync from the other end if it goes
higher in the hierarchical chain. If the port goes to a box consider
lower in the chain, then the distant
On Sat, 20 Nov 2004, Wilson Pickett wrote:
Here's my current plan:
snip
Sounds like a plan?
You asked for advice, here comes some that few will approve of :)
FWIW I tried to get gnophone running and got no further than you did.
What struck me though was that I have a very linux wise
On Thu, 18 Nov 2004, Kevin P. Fleming wrote:
So, I ask again: given the choice between a sub-$100 16-port full-duplex
100Mb switch and external power supplies, and an over-$1000 12-port
switch with internal power supply, which do you think is a better value
for a small LAN? I can buy $20
On Wed, 17 Nov 2004, Noah Miller wrote:
I'm ordering some more phones - I have the Polycom IP 500's now and I
like them. I need some less expensive phones, and I'd like to stay
with all Polycoms for ease of administration. I've heard, though, that
the IP 300's don't support PoE even
On Mon, 8 Nov 2004, Rich Adamson wrote:
That's not right.
New phones come loaded with the current relevant firmware.
Upgraded f/w is only available to/from certified resellers.
Or look on the wiki for where it is freely available.
The two new 500's that were purchased from a Polycom
On Fri, 12 Nov 2004, Jeremy McNamara wrote:
Here is a selected portion of the strings output:
[EMAIL PROTECTED] ~]# strings voipgw | grep Mark
Written by Mark Spencer [EMAIL PROTECTED]
[EMAIL PROTECTED] ~]#strings voipgw | grep CVS
Asterisk CVS-05/30/03-20:39:27 built by [EMAIL PROTECTED]
On Tue, 2 Nov 2004, Michael Giagnocavo wrote:
Yes, there is network support.
I tried it out, but the voice quality seemed quite choppy (local machine,
P4
3GHz). Not sure if it'd actually work for any near-production scenarios.
-Michael
Being partly responsible for AstWind, the
On Wed, 3 Nov 2004, Rich Adamson wrote:
So, while I've posted with respect to FXOs previously, I must ask
againwhat FXO interface device can anyone recommend from real
experience?
I'd have to agree with you 110%; exactly the same issues here over the
past year. I've spent more money
Hello,
I have just picked up a pair of SoundPoint IP 300 phones for
testing purposes, and they work great. Really good quality units for the
price. I only have two complaints/issues with them;
1. The $40 additional cable for POE to work. That sucks.
2. I can't for the life of me figure
On Fri, 5 Nov 2004, Michael Giagnocavo wrote:
CoLinux works great for IAX to IAX or SIP to IAX where no Disk Access is
taking place.
Thanks for clearing that up. I had been using it for IVRs. So if I
created a
RAM disk for CoLinux and booted it from there... that might work?
You
On Fri, 5 Nov 2004, Greg Boehnlein wrote:
Hello,
I have just picked up a pair of SoundPoint IP 300 phones for
testing purposes, and they work great. Really good quality units for the
price. I only have two complaints/issues with them;
1. The $40 additional cable for POE to work
On Fri, 29 Oct 2004, Michael Giagnocavo wrote:
The only thing wrong with RedHat as far as asterisk is concerned is that
they do something goofy with their kernels and all you need do is recompile
a kernel from source. IMHO, you should always compile a kernel for your
specific hardware.
Hello,
I am having a hard time sending a Flash Hook via my Analog -
SPA-2000 - X100P - POTS connection. Anyone have any suggestions?
When I hit flash-hook on the Analog phone the Sipura intercepts it and
puts the caller on hold. I have no way of sending a flash-hook out the
X100P to
On Sun, 24 Oct 2004, Joe Greco wrote:
I know she works at Digium but they probably go down the street to a real
sound stage to do the recordings via 3rd party.
A sound stage is a facility used to create and process professional
recordings. They can be used by anyone employed by an
On Mon, 18 Oct 2004, Claus Lavdal wrote:
I would be very interested in the script that allow me to use
saft_asterisk and non-root user on suse 9.1.
Regards Claus
Claus,
Sorry it took a couple of days to get this information back to
you, but I wanted to let you know that you can
On Tue, 19 Oct 2004, Olle E. Johansson wrote:
I wish it were in v1.0 as well. Would creating a patch for 1.0 be pretty
simple, or do the code changes run deep?
It will eventuall get into a release.
But please, as a community, we have to refrain from temptation of adding
new functions to
On Tue, 12 Oct 2004, Geoff Nordli wrote:
Is this where we get to vote for our favorite router software? I choose
Bering-uClibc
(http://leaf.sourceforge.net/mod.php?mod=userpagemenu=910page_id=36). It
comes with a ton of packages, and you can easily configure it to boot from
HDD, or Compact
On Mon, 11 Oct 2004, Pete Brown wrote:
Greg,
Which kernel are you using? I have two machines at home and the zaptel
kernel module only runs properly on one of them...
The P-3 box worked...
kernel-2.4.20-30.9.i686.rpm
RH73 Zaptel Modules are built against:
2.4.20_28.7.i386
RH9 Zaptel
On Mon, 11 Oct 2004, Mark Phillips wrote:
I had a rather unpleasant bait and switch episode with Atacomm today.
They advertise on their website (and indeed quoted me for) the Snom 200
for $269 which, when I came to place an order for 15 of them, they
didn't have but would like to replace with
On Mon, 11 Oct 2004 [EMAIL PROTECTED] wrote:
Everyone:
We are a Snom authorized reseller and the problem with the Snom 200 is the
fact that Snom has EOL that model. It is being replaced with the Snom 190.
The reason there are no Snom 200's is these unit were taken out of
production
On Tue, 12 Oct 2004, Matthew Boehm wrote:
Switching to DSL would require me to get a phone line, which kinda defeats
the purpose of doing VoIP. =)
Matthew
Matthew, for unparalelled hackability try the Linksys WRT54GS. It runs
Linux, and supports QOS if you use the Sveasoft
On Sat, 25 Sep 2004, Florin Andrei wrote:
On Fri, 2004-09-24 at 05:47, Greg Boehnlein wrote:
Anyone else having the problems that Gary is reporting?
Um, well, not really. I'm rebuilding your package on Fedora 2 (kernel
2.6) and i had to add a linux 26 at the end of the make line
On Sat, 25 Sep 2004, David Troy wrote:
[deleted[
I had a need for a much simpler proxy than his op_server.pl; to meet my
need I re-worked and simplified his code. See below for this simplified
proxy:
http://www.popvox.com/simpleproxy.pl
Hehehehe.. I mentioned this in the Developer's
On Fri, 24 Sep 2004, Greg Boehnlein wrote:
On Thu, 23 Sep 2004, Gary Carr wrote:
The RPMs had errors for me
After installing RPMS and running modprobe zaptel I get
/lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol
register_chrdev_R07a6f6f0
/lib/modules/2.4.20-31.9
Boehnlein
Sent: Friday, September 24, 2004 8:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.0 RPMS RH73 and RH9
On Fri, 24 Sep 2004, Greg Boehnlein wrote:
On Thu, 23 Sep 2004, Gary Carr wrote:
The RPMs had errors for me
Hello,
Please be conscious of Digium's bandwidth and use a Mirror when
downloading 1.0. I have mirrored the tarballs at:
ftp://ftp.nacs.net/asterisk/
Direct links:
ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz
ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz
Hello,
Straight from the floor of Astricon 2004, I am happy to release my
updated Asterisk 1.0 RPMS for RedHat 7.3 and RedHat 9.0 platform.
Current Release
---
asterisk-1.0-0
libpri-1.0-0
zaptel-1.0-0
kernel-module-zaptel-1.0-0
RedHat 7.3
--
On Thu, 23 Sep 2004, Gary Carr wrote:
The RPMs had errors for me
After installing RPMS and running modprobe zaptel I get
/lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol
register_chrdev_R07a6f6f0
/lib/modules/2.4.20-31.9/misc/zaptel.o: unresolved symbol
On Sat, 18 Sep 2004, Bill Seddon wrote:
I've spent a couple of evenings writing a presence utility in C# so that a
window, listing the currently registered SIP phones, can be displayed and a
user can see who is on the phone and who is not. It uses the Manager API
and works well, updating the
On Thu, 9 Sep 2004, hank smith wrote:
I have a SiS 900 PCI Fast Ethernet Adapter what do I put in there or is that
what I put in the xml file?
Go read: http://www.colinux.org/wiki/index.php/coLinuxNetworking
Specifically:
If in doubt, the name of the card can be found in colinux-daemon
On Thu, 9 Sep 2004, hank smith wrote:
is there going to be a gui for co linux and astwind?
No. AstWind is just a Debian GNU Linux distribution with a precompiled
Asterisk installation running under a CoLinux kernel.
I will have to see if either there is going to be a gui or if yasr a screen
On Fri, 10 Sep 2004, Benjamin on Asterisk Mailing Lists wrote:
However, you could use VMware on an Intel notebook to run both Windoze
and Linux concurrently. This wouldn't be ideal for a real PBX for
performance reasons, but since all you are going to use Asterisk for
is to be a gateway for
On Wed, 8 Sep 2004, hank smith wrote:
hello I am fitteling with the astwind-installer-0.1.1.exe asterisk for
windows and am having trouble getting the thing to connect to the meers
to download the updates and stuff. I looked at the wiki and set up
networking and stuff with no success, has
On 9 Sep 2004, khurram bhatti wrote:
Well I wanted to test astwind and consulted * person
he gave me this comment
lord help us all ... why would you want to simulate a linux system on
top of a windows system in the first place?
It's not a simulated linux system. CoLinux is a kernel that
On Wed, 8 Sep 2004, Chris HARIGA wrote:
I make it work!!
My Astwind is up and running!
Now is 11:53 PM and I'm going to bed. Tomorrow morning I will post how I fix
the Ethernet connection.
I bet you followed the following directions! ;)
From:
On Thu, 9 Sep 2004, Chris HARIGA wrote:
[SNIP]
!-- This allows you to modify networking parameters, see the README
or website for more information --
network index=0 name=Intel(R) PRO/100+ Alert on LAN* Management
Adapter type=bridged /
/colinux
Yep.. Don't you hate the
On Thu, 9 Sep 2004, arsal siddiqui wrote:
dear khurram,
i need to know the price for x100p. i've emailed convergence.com.pk
and never get a reply. If you could help me in this regards, i'll be
greatful. I need to know the price.
send me an email off the list. if you can help me in
On Thu, 9 Sep 2004, Martin Mielke wrote:
Hi all,
due to the rather big email traffic regarding this issue, I decided to
publish the script so people can download it at their own risk... :-)
Please, visit:
http://www.leals.com/~mm/asterisk
for further information.
Regards,
On Thu, 9 Sep 2004, Matt - Telcom Products wrote:
Hello,
Does anyone know how to conference a call on the SNOM 200 phone?
Whenever I push the cnf/tran button it just hangs up on the active call.
The manual says you have to push the cnf function key but it doesn't
appear in the lcd on my
On Wed, 1 Sep 2004, Jay Milk wrote:
Hello All,
My asterisk installation has now been running for over two months
without a hitch, and I've decided it's time to move things around a bit.
It's currently installed on a 2.7GHz Celeron under RH9 installed on a
10GB leftover drive. Thanks to
On Fri, 27 Aug 2004, Andrew Brown wrote:
When using the callback feature on agents I notice that when the queue calls
one of the agents and the agent picks up the call they hear nothing until
pressing the # to accept the call.
Only then does my announcement play back to the agent after
On Sat, 28 Aug 2004, Michael Welter wrote:
Since the Cisco 79XX phones preceded the PoE standard, they are
different--polarity is reversed.
IANAE, but as I understand the PoE devices, there are two types--one
always applies -48VDC to the brown pair while the other senses (as per
the PoE
On Sat, 28 Aug 2004, Michael Welter wrote:
Greg Boehnlein wrote:
On Sat, 28 Aug 2004, Michael Welter wrote:
Since the Cisco 79XX phones preceded the PoE standard, they are
different--polarity is reversed.
IANAE, but as I understand the PoE devices, there are two types--one
always
On Mon, 23 Aug 2004, William Boehlke wrote:
Consider hot swappable SCSI RAID 1 instead of IDE. You'll appreciate it
every couple of years when you lose a disk but the PBX stays up.
I hot swap on my 3Ware IDE RAID card all the time.
--
Vice President of N2Net, a New Age Consulting
On Sat, 21 Aug 2004, Ed Devine wrote:
I've got a Dual Xeon system Digium Quad T1 and an Idle T1 circuit that
I can experiment with. I've been wanting to use Redhat with software
Raid 1 on an Asterisk server.
Has anyone had any experience with software raid and Asterisk? Also, if
the
On Sat, 21 Aug 2004, Lyle Giese wrote:
I used software raids and want to get away from them. I really really
like Microlite's BackupEdge tape backup software. BackupEdge does not
work with a software raid, only a hardware raid.
The second kicker was that the Promise (or any other)
On Thu, 12 Aug 2004, Olle E. Johansson wrote:
If you look into the download areas, you'll find Asterisk 1.0 release candidate
two...
Find the mirrors on the link below, they'll update during the day if they don't
already
have RC2. Please don't hit the Digium FTP-server, since development
On Fri, 6 Aug 2004, Joshua M. Thompson wrote:
On Fri, 2004-08-06 at 10:16 +, Ryan Courtnage wrote:
Are you using a newer zaptel build? If so, make sure you didn't get caught by
bug #2218. wcfxs had an error introduced this week, which is now resolved in
HEAD.
It's possible,
On Fri, 6 Aug 2004, Robert Hajime Lanning wrote:
Well, the G729 codec would need to be recompiled to take advantage
of the 64bit arch.
I've got a Dual Opteron box running Gento x86_64. If someone could point
me in the right direction and provide some training on how to test this,
I'd be
On Tue, 3 Aug 2004, Tom wrote:
At 07:08 PM 8/3/2004, you wrote:
That sigh will turn to cursing after a couple of months. We currently use
Rodopi, have for 10 years but the inflexability is too much to deal with
anymore so we are moving away from it.
To what? I am also a cursed Rodopi
On Wed, 4 Aug 2004, Jeremy McNamara wrote:
Greg Boehnlein wrote:
We use Platypus from Boardtown, which was just acquired by Tucows.
Although it has it's quirks, having seen Rodopi, Emerald, Prism and
ISPEasy in action, I'll take Platypus ANY day!
I have a method for hacking VoIP
Hello,
I am attempting to get the asttapi driver working on Windows XP
Professional, and am running into some strange problems. I've combed the
Web and the Wiki for information on debugging the application to see if I
can solve my issue, but nothing is helping me. I have tried the
On Wed, 28 Jul 2004, James H. Cloos Jr. wrote:
Has anyone does this with *?
Ie, ask for the caller's name and provide that to the callee before
bridging?
For calls to an extension, it should be doable via the dialplan. For
calls to queues, some changes would be required to app_queue.c
On Thu, 29 Jul 2004, Mike Machado wrote:
Another thing you could do is use a regular phone to call into a DID and
enter the conference, then everybody can join that conference and listen. No
bandwidth required, just a phone call to the distributor's Asterisk server.
Then just keep that
On Thu, 29 Jul 2004, Steve Woolley wrote:
I thought the interesting exercise would be to use asterisk for the
task. Couldn't we use a kind of distributed conference call where a few
key select/high bandwidth asterisk servers form the main conference and
then have multiple layers of
On Wed, 28 Jul 2004, Philipp von Klitzing wrote:
Hi!
When TRANSFERING a call to another extension, if you enter an invalid
extension, (I.e. Hit TRANSFER, then dial erroneous number.. SEND, Congestion
tone, Hang-up, go off-hook.. Try different solutions to try to get call
back But
On Wed, 26 May 2004, Steve Totaro wrote:
Fedora 2 core put up a little fight but not much.
I installed Gentoo on my Dual Opteron box w/ a completely native x86_64
2.6.7 kernel, and was able to get my X101P working w/ Zaptel by simply
doing make linux26. Works great in 64 bit mode.
-
On Sun, 25 Jul 2004, Jay Milk wrote:
Surely you mean grammar? Sorry, I just had to point that out :)
Personally, I'd take issue with the title -- if you need to do a
small-office setup by-the-book, then chances are you're not
resourceful enough to find the required information online --
On Wed, 21 Jul 2004, Steven Sokol wrote:
Has anyone really looked at the costs for Astricon. But the hotel costs.
$111.00 USD per night.. come on guys give me a break. I will not be
staying at that hotel. I can rent a car and stay near the air port for
almost half that.
In
On Thu, 22 Jul 2004, Mike Reed wrote:
If you haven't been to Atlanta before, their public transportation is
decent, too. Train runs right to the airport, and straight into town.
Last few conferences I've had there, the train elt me off within 2
blocks of my hotel.
That is what I was hoping
On Wed, 21 Jul 2004, Jay Milk wrote:
Just received:
Cognigen is very proud to announce the official launch
of Broadvox Direct, a new VOIP service.
Broadvox Direct offers unlimited calling plans to
anywhere in the US and Canada for a low monthly payment starting at
$29.95 and basic
Hello all,
Hot on the announcement this morning by Mark, I have updated and
rebuilt new Asterisk RPMS for RedHat 7.3, 8, 9 and Fedora Core 1. Please
feel free to install these and beat them up. The usual disclaimer
applies.. These haven't been tested, not reccomended for production use,
On Wed, 7 Jul 2004, brian wrote:
Anyone with a PRI/ISDN line can set callerid to anything... Not just voip,
not just asterisk. Come on guys.
bkw
Yes, but the Telco has the ability to either pass or deny that. In my X/O
PRI configuration, I can only set the CallerID to a number within the
On Thu, 8 Jul 2004, Neil Cherry wrote:
Stephen R. Besch wrote:
The most recent version of GSConfigure is available at
www.buffalo.edu/~sbesch Several serious bugs that kept the program from
getting started have been ferreted out and corrected with the help of
Bruce Komito. The
Howdy,
I just did an apt-get dist-upgrade on my Debian unstable box,
and noticed that the Asterisk version appears to be 1.0-1 in the unstable
tree. I KNOW that 1.0 hasn't been released yet, so I am wondering who is
responsible for the Debian packages? This will be VERY VERY confusing
On Sun, 27 Jun 2004, Max Lock wrote:
Hi Folks,
Since there isn't a grandstream forum AFAIK I guess someone here may be able to
shed some light on this. Apologies if this is viewed as offtopic..
I think I may have killed the firmware on my Grandstream Budgetone 101. I found a
On 29 Jun 2004, Adnan Shah wrote:
Hi !
I need a solution to park incoming calls
to an extension of my choice where a special
announcement is played, park subsequent calls
to specific pools so that they listen to announcements
of my choice.
any ideas ?
Put up a bounty on it of a
On Mon, 31 May 2004, Tony Hoyle wrote:
Stefan de Konink wrote:
Is Asterisk not a *little bit* too much for that processor? SER could be a
better choice?
The asterisk binary alone is larger than the total flash ram space on the linksys.
I really doubt it's going to work
That
On Mon, 31 May 2004, usedcanon wrote:
I have used with Athlon 64, but noth opteron. Can imagine it being much
different though.
I'll let you know in a couple of weeks when my Dual Opteron workstation is
finished.
--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
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