Hi, List
The Gentoo portage tree only include 0.9.0, it seems no upgrade for long time.
Do you know someone have the 1.0.1 ebuild version?
--
Jacky
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Justin wrote:
Olle,
That's a great start but as the documentation states:
NOTE: this requires substantial work to be sure that Asterisk's
environment has permission to write the files required for its
operation, including logs, its comm socket, the asterisk database, etc.
Can that be made ea
Call Manager's biggest problem is that it has an embarrasing lack of
features, and is not considered reliable at all.
Porting it to a better operating system is certainly a good idea, but
the PBX part of it is totally amateurish.
Asterisk trumps ALL PBXs on flexibility, but Call Manager is famous
> On Wed, 20 Oct 2004 17:27:38 -0500, Henry Devito <[EMAIL PROTECTED]> wrote:
>
> > HI I am in the US and have a customer using * in the US they just acquired
> > a call center in India. Does anyone know if I can legally sell/ship
> > Grandstream IP phones and IAXy's to India?
>
> I am sure you
Hi All !
First I was having trouble using attended call transfer using asterisk but
thatnks to you guys I was able to make it work by adding 't' in options
and applying the patch. Now I am using SER along with asterisk. SER works
as SIP proxy and Asterisk performs all the necessary PBX functio
> Best value in gig switches right now is Dell. Go to Dell Small
> Business and keep an eye out for some deals. They have a pretty good
> one going on now for their 2000 series.
>
> http://www1.us.dell.com/content/products/compare.aspx/2000_workgroup_gig?c=us&cs=04&l=en&s=bsd
>
> *Not affiliate
Nicolas
just playing with the new ming version of the your panel
& getting some version grief ;(
could you post what versions & urls
libpng
libungif
..and the perl version of
ming
Data-TemporaryBag
and but not least SWF-File
mayebe best to stuff in the README or sumfin
cheers nice work
_
Magnus:
> Yeah, thats what I figured, BUT, if you transfer an incoming
> call to another internal user, music on hold switches to
> INTERNAL, and if the 2nd agent does a another transfer, the
> incoming call gets INTERNAL music.
Only if your dial plan is set up that way. It is possible to mak
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Christopher L. Wade
> Sent: October 20, 2004 5:46 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] IP Phone that OFFICIALLY
> support Asterisk
>
>
Okay so I have a couple Voip accounts. I want my dialplan to prefer the
first until I get to some threshold on minutes used for the month then
prefer the other. (read: one has free minutes while other's per-minute
charge is lower) Is there a neat trick to handle this or do I need to
spin up an A
The only TIE lines a 280 supports are dumb tie lines. I am a Toshiba dealer
Email me off list for details, yes I did set this up in the lab.
-Original Message-
From: Brian Roy [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 20, 2004 9:17 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-
Brian Roy wrote:
Bleh, what "group" said this? The 3Com group? Dell is the WalMart of
the hardware world. Their pricing is better because they build
efficiencies. I have 33XX 54XX and I just bought my first 6024 Layer 3
QOS ready switch. These things are nothing but Ciscos in sheep's
clothing. They
This flashing is an indicator for a damaged firmware in your phone.
Maybe an interrupted TFTP-Download when powered up or just a "wrong"
firmware.
dean collins schrieb:
Does anyone know what it
means when a grandstream flashes
the red key light 5 times repeatedly in cycles? I g
Steve:
You wrote:
An operator could take control of the call and reroute it, but I'm not sure
how you would alert the operator and get them involved.
That gaves me an idea. The asterisk box will have also 12 analog FXO
signalled lines, so if an operator can reroute a call, Asterisk can act as
an
Does anyone know what it means when a grandstream flashes
the red key light 5 times repeatedly in cycles? I got a new handset delivered
to me today, powered up fine until I tried to access it via the web interface
using the password admin and then it rebooted with the lcd never displaying
a
Steve:
Thanks about the explanation. I'm rather new to all of this digital
telephony world. I'm a computer networks guy :)
If I understood well, the transfer limitation isn´t a MFC/R2 one, but a PSTN
one?. Can I transfer calls using the PBX call control even in R2 if the PBX
support it? Flash di
David Hajek wrote:
Hello,
I need to include many contexts based on time and date. But I have a so
called "midnight" issue. I want to include context when time is between
10pm,Oct 21 - 3am,Oct 22. Is it possible to write it using one line?
include => context1|22:00-00:00|*|21|Oct
include => context1
Incoming calls without calling id go the the "s" extension.
extensions.conf
[incoming]
exten => s,1,Hangup
zapata.conf
context=incoming
channel => 1
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Alex van Es
> Sent: Thursday, 21 October 2004 1
Hi all,
Sorry if this has been answered previously, but I have not had any
luck trying to find it.
I am trying to connect my Asterisk server (1.0 stable, Fedora Core 2,
kernel 2.6.8-1.521) to connect to a gateway that can only support
H323. I have installed the asterisk-oh323 channel driver (vers
Bleh, what "group" said this? The 3Com group? Dell is the WalMart of
the hardware world. Their pricing is better because they build
efficiencies. I have 33XX 54XX and I just bought my first 6024 Layer 3
QOS ready switch. These things are nothing but Ciscos in sheep's
clothing. They have been rock s
Jon Radon said:
> Best value in gig switches right now is Dell.
Another data point for ya,
Installed a NetGear GS724T at a site just today. Got it for about $650
from buy.com. 24-port gig, 2 mini GBIC, trunking, tagging, vlans,
mirroring, web management, QOS, SNMP, etc. It's not a "managaged"
Hi
will check the relay part, as for the loop not happening I pasted wrong
lune, basically where the EXTEN part is I had hardcoded a username which
existed in my DB, and that got now loop...but will check again.
What I am trying to set up, is to get SER to register all the calls, and
pass these
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Edwin Quijada
> Sent: October 20, 2004 3:26 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Newbie with new Project VOIp
>
>
> Hi!
> I am a newbie in VoIp. Looking for in the net I get thi
I wanted to get some feedback on an issue that has
popped up since we brought up a new Asterisk server.
We are running v1.0.1 on a dual Xeon 3Ghz
processor server with 1GB RAM. Our install and configuration
is
practially identical to our other server
except that we have installed a T100P
On Wed, 2004-10-20 at 17:27 -0400, Jim Van Meggelen wrote:
>
> And as for Call Manager? I predict that they will be officially
> Asterisk-compliant in . . . hmmm . . . I'll say roughly five years or
> so. Possibly far sooner if they yank their heads out of their asses and
> grab a clue. Asterisk is
On Wed, 20 Oct 2004, Theo Zourzouvillys wrote:
after a couple of days work banging my head against the wall (bloody
standards my arse), i've got chan_bluetooth to a point where it's
starting to function - certianly more than just proof of concept now.
Just to note it here too (along with the chan
Hello Ron Ramos
With the power of bash it's easy
copy this code to a file name it createsip.sh
#!/bin/bash
for ((i=2000 ; i < 8000; i++ )); do
echo "[$i]"
echo "secret=$i"
echo "type=friend"
echo "username=$i"
echo ""
done
---
Hi All,
How can I be able to define multiple SIP extensions?
Do I have to define each extensions on sip.conf?
For example, extension 2000-8000, do I have to define it one by one
on sip.conf?
[2000]
secret=2000
type=friend
username=user2000
..
..
[2001]
secret=2001
type=friend
username=user200
Wed, 20 Oct 2004 15:47:59 -0500, Henry Devito <[EMAIL PROTECTED]> wrote:
> Where can I buy the act phones?
Since we are helping them with some stuff we buy them directly,
although we wouldn't normally qualify to buy directly as our order
quantities are low. I don't know who their resellers are in
Lol, this email was sitting in my inbox as I was reading this.
It's not about bandwidth! Learn why application performance can't be
solved with bandwidth or compression.
October 21 @ 4 p.m. Eastern/1 p.m. Pacific
Duration: 30 minutes
Register to Attend
You might want to look at the Dial command and check how many leading digits
you are stripping from the number that you are dialing.
It caught me today on a mindless cut and paste job from an existing working
dial plan entry.
--Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMA
Hi Everyone
We would like to connect our Splicecom Maximiser PBX to our Asterisk box
via H323 so that we can send our US calls via a low cost carrier (e.g.
Broadvoice).
Has anyone managed to do this in the past (I remember seeing some
companies also worked with this system in the UK).
The Maxi
On Oct 20, 2004, at 4:17 PM, Jay Wilton wrote:
What is the most important feature for VoIP quality:
latency, qos, vlan? I'm leaning towards least latency with
qos and/or vlans at the linux router. Might be my best
shot for an inexpensive gig switch ($100).
I have only seen the qos (802.1p) in the
I just installed it 2 days ago and I was looking for a long time to get
thes problem solved.
Be aware off creating the Mysql database the astcc.cgi script will do.
But I disable the create lines in the perl script.
The astcc.cgi script will endles wait.
Good luck
_
Hello .
I have a 800 tollfree trhough iax to my * server.
If I phone to 800 number to the * machine to a sip phone everything is
okay.
exten => 8,1,Dial(SIP/12345678,20)
-- Accepting AUTHENTICATED call from xx.xx.xx.xx, requested format =
4, actual format = 4
If I change my extention
Guillermo Freige wrote:
Steve:
This means the only way to use Transfer (or Hook and DTMFSend) in a E1
is using it as a channel bank trunk using FXO signaling?. I really
need to free those channels.
FXO signaling cannot reroute the call. You are relying on * to do that
work, as an extension of th
I found this patch a few days ago (on a mailing list), and patched it
against the latest cvs which I downloaded for app conference. With
these changes I believe everything compiled fine no other tweaks
required other then the include dir for asterisk in the make file.
On a side note, id like to s
Yeah I have callerid=asreceived in my zapata.conf still nothing
unfortunately.
I get that when the calling party has caller id blocked on their end.
--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
___
Asterisk-Users maili
On Wed, 20 Oct 2004 17:27:38 -0500, Henry Devito <[EMAIL PROTECTED]> wrote:
> HI I am in the US and have a customer using * in the US they just acquired
> a call center in India. Does anyone know if I can legally sell/ship
> Grandstream IP phones and IAXy's to India?
I am sure you can legally s
Nahuel Alejandro Ramos wrote:
Hi everyone,
I am looking for a prepaid billing solution for my VoIP. I have
already install SER but it does not support a statefull tracking of
the call, so it is dificult to hang up a call when it has zero credit.
Posting messages on Serusers maillist there are a lo
All
I am using the outgoing spool to start the call then run my AGI once
the call is placed. All that works.
The problem I notices was that if the number I am calling is busy
I dont get any notification about that. I can sort of understand the
line cant tell me when the user picked up but how do I
--- Kristian Kielhofner <[EMAIL PROTECTED]> wrote:
> Scott Laird wrote:
>
> >
> > On Oct 20, 2004, at 3:38 PM, Michael Welter wrote:
> >
> >> Kristian Kielhofner wrote:
> >>
> >>> Michael Welter wrote:
> >>>
> Is 802.1p what we need for voice traffic? QoS at
> the MAC level?
>
> >>>
Hi Umar,
[super big snip]
> I want users to be able to do something simillar using there handsets. The
> reason I asked is that I am assuming that Nicholas is transferring the calls
> to a meetme conference, using the management api and then landing the third
> person in the same conference in lis
I don't really like swapping binaries but... I have an
app_conference.so binary file I could send to you if you like. It is
working on the latest stable cvs as of a few days ago. If you would
like it, please let me know and I will get it available.
Darren Wiebe
[EMAIL PROTECTED]
Steve Kann w
Hi,
I have installed asterisk with 1 X100P card and one grandstream ip phone.
When I pickup the phone, I have no dialtone, and unable to dial out.
What configuration files and entry lines, that I should review to get
phone working.
I can dial into asterisk pbx and dial grandstream phone exten
Scott Laird wrote:
On Oct 20, 2004, at 3:38 PM, Michael Welter wrote:
Kristian Kielhofner wrote:
Michael Welter wrote:
Is 802.1p what we need for voice traffic? QoS at the MAC level?
802.1q and 802.1p are preferred.
The reason I asked is because 802.1q isn't mentioned in the product
literature.
Michael Welter wrote:
Kristian Kielhofner wrote:
Michael Welter wrote:
Is 802.1p what we need for voice traffic? QoS at the MAC level?
802.1q and 802.1p are preferred.
The reason I asked is because 802.1q isn't mentioned in the product
literature.
For Dell switches? 802.1q is VLAN support. I
On Oct 20, 2004, at 3:38 PM, Michael Welter wrote:
Kristian Kielhofner wrote:
Michael Welter wrote:
Is 802.1p what we need for voice traffic? QoS at the MAC level?
802.1q and 802.1p are preferred.
The reason I asked is because 802.1q isn't mentioned in the product
literature.
.1q is VLAN trunkin
Kristian Kielhofner wrote:
Michael Welter wrote:
Is 802.1p what we need for voice traffic? QoS at the MAC level?
802.1q and 802.1p are preferred.
The reason I asked is because 802.1q isn't mentioned in the product
literature.
___
Asterisk-Users mailin
Michael Welter wrote:
Is 802.1p what we need for voice traffic? QoS at the MAC level?
802.1q and 802.1p are preferred.
--
Kristian Kielhofner
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To U
HI I am in the US and have a customer using * in the US they
just acquired a call center in India. Does anyone know if I can legally
sell/ship Grandstream IP phones and IAXy’s to India?
Thanks
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[EMAIL PRO
I think I saw someone suggesting there are problems with FWD IAX
registrations. In case Ed Guy or one of the other FWD people is watching
the list, we're also having horrible problems receiving and sending calls
via FWD.
The symptom is that the registration will take forever to succeed and when
i
Every time I asterisk to retrieve voice mail, or dial to
the menu extension there is choppy sound coming out. When that happens asterisk
reports Received bad packet with bad udp checksum. I can have conversation to
other people just fine, but any voice exchange with asterisk is horrible. W
Shawn Dillon wrote:
Thanks to all who have
helped me build and test out Asterisk
installation thus far. I needed to move my * installation to a new box
, due to
the fact my test machine would not support PCI 2.2 ( which I am told is
required to use my TDM11B).
I have * up
Jim Van Meggelen wrote:
Can you imagine!?! That would be the most brillant thing they could do!
They should also open up their platform so people can port Linux, BSD
and what-all-else to it. Oh, and drop the price a bit while their at it!
The whole Linksys/Linux/Cisco thing really fascinates me. Th
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bill Seddon
Sent: 20 October 2004 22:06
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] manager interface to barge
-Original Message-
From: [EMAIL PROTECT
Hello,
I need to include many contexts based on time and date. But I have a so
called "midnight" issue. I want to include context when time is between
10pm,Oct 21 - 3am,Oct 22. Is it possible to write it using one line?
include => context1|22:00-00:00|*|21|Oct
include => context1|00:00-03:00|*|22
Jon Radon wrote:
Best value in gig switches right now is Dell. Go to Dell Small
Business and keep an eye out for some deals. They have a pretty good
one going on now for their 2000 series.
http://www1.us.dell.com/content/products/compare.aspx/2000_workgroup_gig?c=us&cs=04&l=en&s=bsd
The Smc 8508T
Sorry for the OT, just don't know where else to ask :(
What's up with Chagres.net?
They used to be somewhat active on the list. I purchased a GS 102 from
them. It died. I've been trying for about a month to get an RMA from them.
None of the extensions in their menu are ever answered -- they all s
Hi;
I need to sell the following cards if any
interested:
1) Dialogic Telephoney card of 15 ports,
model:Dialogic D/160SC-LS, ISA slot.
2) Truck Card of mdeol: TP100B-32XXSP, PCI
slot.
Price: 1750$
Regards
Bilal
___
Asterisk-Users mailing li
Alex van Es wrote:
>
> Lance,
>
> I noticed my X100P is answering all calls that come in.. do you know
> how I can set up an extension for it to
> stop answering calls?
>
I'm not 100% sure about this but I would think you'd basically have to
unload the driver to get the card not to answer at
>
>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of
>> Christopher L. Wade
>> Sent: October 20, 2004 2:35 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [Asterisk-Users] IP Phone that OFFICIALLY
>> support Aste
bra, but when calling to it
> > nothing happens. I would be happy to help debugging and/or enhancing
> > the code :)
> >
> >
> >
> > Stefan de Konink
> > ___
> > Asterisk-Users mailing list
> > [EMAIL
On Oct 20, 2004, at 1:47 PM, Matt Hess wrote:
Remember, you pay for what you get.. especially with Dell networking
equipment. I have heard about several groups who tried the dell
switches only to give up on them because the dell switches just didn't
perform. Yes, price-wise they look good.. but
Thanks to all who have helped me build and test out Asterisk
installation thus far. I needed to move my * installation to a new box , due to
the fact my test machine would not support PCI 2.2 ( which I am told is
required to use my TDM11B).
I have * up and running and I am attempting t
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nicolás
Gudiño
Sent: 20 October 2004 18:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] manager interface to barge
Hello,
On Wed, 20 Oct 2004 09:48:43 -0600, TELU
Marcelo Pacheco wrote:
Em Qua 20 Out 2004 15:14, Andrew Edmond escreveu:
Asterisk Community --
I'm looking for a way to gracefully shutdown asterisk at least once a
day and bring it back online. I'm using Gentoo Linux and using
safe_asterisk from /etc/init.d/asterisk.
#!/bin/bash
export PATH=/sbi
Remember, you pay for what you get.. especially with Dell networking
equipment. I have heard about several groups who tried the dell switches
only to give up on them because the dell switches just didn't perform.
Yes, price-wise they look good.. but as far as performance goes.. (that
is assumin
Where can I buy the act phones? I went to their website and they look like
decent phones.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin on
Asterisk Mailing Lists
Sent: Wednesday, October 20, 2004 1:53 PM
To: Asterisk Users Mailing List - Non-Co
Best value in gig switches right now is Dell. Go to Dell Small
Business and keep an eye out for some deals. They have a pretty good
one going on now for their 2000 series.
http://www1.us.dell.com/content/products/compare.aspx/2000_workgroup_gig?c=us&cs=04&l=en&s=bsd
*Not affiliated with dell..
Thanks for reply. Yes i am getting audio. It hangs-up automaticly after
10 secs, or the line goes down. Softphone has the line still open
though.
I dont get this 404 anymore, it was just before the missing canreinvite=
-Original Message-
Cinoss,
Are you getting audio during the call?
Michael & Stephen;
I have been running GS BT101's for the past few months in a fixed IP
arrangement and have not had a problem with the registration process.
Budgetones seem very reliable. I have the phones configured to do
registration and expire every minute. I have SIP user ID and
Authe
On Wed, 20 Oct 2004 19:26:12 +, Edwin Quijada
<[EMAIL PROTECTED]> wrote:
> 1-Using * can integrate VOIP phone with analog phone and what that I need?
analog telephone adapter (ATA) or PCI interface card with FXS ports
> 2-Which VOIP phones Can I use with *?
almost any phone that speaks any o
Not claiming that this is a complete list, but I can state with certainty
that the following companies support Asterisk users and integrators:
Grandstream
SNOM
Uniden
IpDialog
Pulver Innovations
Cisco and Zultys don't officially support it, but seem to "tolerate"
Asterisk. The only company that
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nicolás
Gudiño
Sent: 20 October 2004 18:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] manager interface to barge
Hello,
On Wed, 20 Oct 2004 09:48:43 -0600, TELU
On Wed, Oct 20, 2004 at 07:26:12PM +, Edwin Quijada said:
> Hi!
> I am a newbie in VoIp. Looking for in the net I get this product to work
> for Linux, now I have a few questions
> I have a customer that wants implement VoIP using phones VOiP and analog
> and integrate it into network voice/
> In addition I don't want to be in a situation like this when we'll
> mention Asterisk and they will immediately reply "... we don't support
> Asterisk..." Hence my inquiry.
I can tell you that Uniden's support/dev are running Asterisk in-house,
and that they test the UIP200 with it.
_
On Wed, 2004-10-20 at 12:52, Benjamin on Asterisk Mailing Lists wrote:
> On Wed, 20 Oct 2004 11:58:39 -0600, Joseph <[EMAIL PROTECTED]> wrote:
> > What IP Phones officially support Asterisk. I know that most of them
> > will work with * but I do not want to support companies that don't
> > support
Hi!
I am a newbie in VoIp. Looking for in the net I get this product to work
for Linux, now I have a few questions
I have a customer that wants implement VoIP using phones VOiP and analog and
integrate it into network voice/data.
1-Using * can integrate VOIP phone with analog phone and what tha
On Wed, Oct 20, 2004 at 01:46:01PM -0400, Stephen R. Besch wrote:
>
> I have never been able to get the Grandstream to register reliably -
> with any version of the firmware.
So you mean you don't use the Grandstreams, then?
> It sounds like in your test with the
> fixed IP, you left the regis
asterisk*CLI> show application UserEvent
asterisk*CLI>
-= Info about application 'UserEvent' =-
[Synopsis]:
Send an arbitrary event to the manager interface
[Description]:
UserEvent(eventname[|body]): Sends an arbitrary event to the
manager interface, with an optional body representing additi
On Wed, 20 Oct 2004 11:58:39 -0600, Joseph <[EMAIL PROTECTED]> wrote:
> What IP Phones officially support Asterisk. I know that most of them
> will work with * but I do not want to support companies that don't
> support OSS
I guess it all depends on your definition of "support OSS". Does it
mean
--On Wednesday, October 20, 2004 09:21 -0500 [EMAIL PROTECTED] wrote:
If someone else doesn't give a better solution, you can try this.
Don't run * as root. chown /dev/zap to say asterisk and run asterisk as
that user. Make sure the ASTVARLIB (/var/lib/asterisk) and spool dirs are
all owned b
Olle,
That's a great start but as the documentation states:
NOTE: this requires substantial work to be sure that Asterisk's
environment has permission to write the files required for its
operation, including logs, its comm socket, the asterisk database, etc.
Can that be made easier or is t
On Wed, 20 Oct 2004 19:41:52 +0200, Alex van Es <[EMAIL PROTECTED]> wrote:
> I noticed my X100P is answering all calls that come in.. do you know
> how I can set up an extension for it to stop answering calls?
what is the context you have assigned to the FXO channel? (check in
/etc/asterisk/zapat
Is there a way to flag the manager API on an event from the dialplan?
db
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Matthew Crocker wrote:
I don't know if Cisco officially supports Asterisk but I know they do
provide funding/programmers for many OSS projects (www.vovida.org VOCAL)
being one of them.
Just wish they'd OS the firmware :)
--
Christopher L. Wade Unistar-Sparco Computers, Inc.
Se
I don't know if Cisco officially supports Asterisk but I know they do
provide funding/programmers for many OSS projects (www.vovida.org
VOCAL) being one of them.
-Matt
On Oct 20, 2004, at 1:58 PM, Joseph wrote:
What IP Phones officially support Asterisk. I know that most of them
will work with
Justin wrote:
It is great that this documentation is out there, and that *
supports this. However I think in an ideal world this would be inherently
supported by * and ideally setup via config file like with apache:
User www
Group www
From the Asterisk man page:
asterisk [ -hfdvVqpRgcin ] [ -
Good luck! Personally I like my cisco 7960's
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Joseph
> Sent: Wednesday, October 20, 2004 12:59 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] IP Phone that OFFICIALLY suppor
On Wed, 20 Oct 2004 18:48:11 +0200, Alex van Es <[EMAIL PROTECTED]> wrote:
> phoneline that goes into the x100p card and on the phone jacket of the
> card I connected a regular pstn phone.
> Does anyone know if any of these following things would be possible
> with my setup;
> - Receive the calleri
On Wed, 20 Oct 2004 11:58:10 -0500, Eric Wieling <[EMAIL PROTECTED]> wrote:
> It does not appear to work for calls from Zap FXS ports but that's the
> only time I've noticed that it doesn't work.
I don't know about Zap FXS ports, because I haven't got any, but I can
tell you that none of the regis
Problem fixed for the Vmail Soft Key
I had the userfrom in the wrong context.
Anyhelp on the hold button is still appreciated.
Good day list,
Need
some assistance in setting up the snom190 with asterisk.
My
voicemail server is a
What IP Phones officially support Asterisk. I know that most of them
will work with * but I do not want to support companies that don't
support OSS
--
#Joseph
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Thanks Senad, but I am looking for free software.
Nahuel Ramos.
On Wed, 20 Oct 2004 18:49:34 +0100, Senad Jordanovic <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Have a look at this for your needs...
>
> www.bicomsystems.com
>
> Regards,
> Senad J
>
>
>
> Nahuel Alejandro Ramos wrote:
> > Hi
Kristian Kielhofner said:
> This is well documented in the wiki and elsewhere.
Yes, I know. I even quoted it in my note ;) What I'm suggesting is that
it should do so by default, not with some additional changes to the
standard installation.
Please don't flame me for this but there are far too
Michael George wrote:
I am having trouble with a Grandstream Budgetone 101. It's at firmware
1.0.5.10 and I'm running * 1.0.0.
I have the phone getting a DHCP address and * expects it to register.
When I reboot the phone it does register just fine. However, after a while *
cannot contact the phon
Marcelo,
Great idea to pull zaptel/etc out of modprobe memory too and restart
EVERTYHING. Great idear.
Andrew
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marcelo
Pacheco
Sent: Wednesday, October 20, 2004 10:32 AM
To: Asterisk Users Mailing List - No
Lance,
I noticed my X100P is answering all calls that come in.. do you know
how I can set up an extension for it to
stop answering calls?
I have tried
exten => Zap/1-1,1,wait(180)
exten => Zap/1-1,2,Answer
exten => Zap/1-1,3,Hangup
But this doesnt seem to work.. anyone any suggestions?
Alex
On 20
Our experience with VoIP has been that while you could use a large ping
packet to give you an idea of your network performance doing a transfer
test of a large amount of data is better.
To do this clear the counters on your switches and then do a transfer of
10 GB of data at several times throug
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