All of those lines going to desktops must be fed from someplace, like
a wiring closet maybe? Why not put your FXO gateway there instead of
at the desktop where your users will break them?
On 2/20/09, Paul Hales pdha...@optusnet.com.au wrote:
Why do you need so many Asterisk installs?
With
Steve, what kind of Avaya system is this? They make several.
On Tue, Aug 5, 2008 at 11:36 AM, Steve Davies [EMAIL PROTECTED] wrote:
Hi,
Sorry this is so long, but I am reasonably desparate.
I am having real fun with hooking an Avaya system to Asterisk using
ISDN30. I have the ISDN
Well, my $21 is still there and all of my calls are being declined.
Over a year ago, I requested a refund and regardless of all promises that I
would receive one, Jed never followed through. I'd use up the credit if the
calls would only complete.
On Sat, Apr 5, 2008 at 1:03 AM, Ira [EMAIL
This may be a good place to start looking:
http://www.atlassound.com/index.cfm
On 2/20/08, Jerry Geis [EMAIL PROTECTED] wrote:
I am looking for an ATA like device but instead of VOIP to analog phone
I want VOIP to low level audio out. Something that looks like a sound card
output.
I know I
I just registered an Avaya 9620 set to my Astlinux system (0.47 - Asterisk
1.2.22), using Avaya SIP Firmware version 2.0.1.34.
Set [EMAIL PROTECTED] in the sip.conf
Found MWI worked immediately. Turned off as expected.
Have Fun!
Tom
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You're looking for Leave Word Calling activation and deactivation.
On 12/28/07, Doug Lytle [EMAIL PROTECTED] wrote:
Henk Dick wrote:
Doug,
Have you checked the feature access code that is defined in the
definity. That is the code that needs to be dialed. I always checked
the codes
I suspect if you remove the callerid entry from this device's
sip.confdefinition things will work better.
On 9/9/07, Apa Minerala [EMAIL PROTECTED] wrote:
I have searched this list and others, and see other pepole having this
issue. However, I have not seen how to fix it.
Sep 6 18:52:36
I'd love to hear about this as well.
On 7/27/07, Derek Fedel [EMAIL PROTECTED] wrote:
Hi all,
I'm new to the list, so I apologize in advance if I'm beating a dead horse
by asking this, but I read somewhere that asterisk 1.4 has MWI working for
Avaya and their rather troublesome SIP
On the other hand, the guy could just be using his work e-mail for personal
interests.
On 7/7/07, Matthew Rubenstein [EMAIL PROTECTED] wrote:
On Sat, 2007-07-07 at 08:39 -0500,
[EMAIL PROTECTED] wrote:
Date: Fri, 06 Jul 2007 12:02:53 -0600
From: Stephen Bosch [EMAIL PROTECTED]
Subject: Re:
exten = _X.,1,
On 6/3/07, BSumrall [EMAIL PROTECTED] wrote:
Understood, it is not the catch all but, what if I am designing a server
that needs to accept calls from 15 or more 1800 numbers?
How would you now channel it to a catch all?
Brad
-Original Message-
From: [EMAIL PROTECTED]
Bilal,
I don't think anyone is telling you that digital phones don't need cards.
I do think they are telling you that NOBODY makes a card that drives digital
phones for use with Asterisk.
Your initial assumption that Digital phones work with Asterisk is untrue.
They can be made (forced?) to
At the very least, he's abusing his customers. Substances? I hadn't
thought of that.
On 4/30/07, Salvatore Giudice [EMAIL PROTECTED]
wrote:
I suspect that Jed has a substance abuse problem and that he may be in
rehab. I don't know for sure of course. This kind of silence is indicative
of
You could also look at Oreka at sourceforge.
On 4/13/07, Matthew J. Roth [EMAIL PROTECTED] wrote:
Savoy, Kevin - Williston, ND wrote:
We are looking at using Asterisk as a call recording server for an
Avaya VoIP S8700 system in a multi-site VoIP Call Center. All calls
will be coming in to
Check out Oreka at sourceforge, too.(aka OrkAudio)
On 2/15/07, Kristian Kielhofner [EMAIL PROTECTED] wrote:
On 2/15/07, Cory Andrews [EMAIL PROTECTED] wrote:
Apologies in advance as this is not directly Asterisk related, however I
thought I might be able to leverage the experience of
Lacy, it appeared to me that he was calling himself an idiot. Thanks for
some of the background on the issue, though.
On 3/27/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
WOW that fixed it! What an Idiot.
I was going somewhere
Balu,
I suspect the author was expressing sarcasm.
On 3/26/07, Balu Raman [EMAIL PROTECTED] wrote:
Can you tell me, why sellvoip rocks ?
I am looking to sign up with someone.
Thanks,
balu raman
On 3/25/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Salvatore Giudice wrote:
Nothing has changed
I'm not surprised.
On 3/25/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Salvatore Giudice wrote:
Nothing has changed in my Asterisk configuration and now outbound US is
getting nothing, but 403's. Anyone else having the same problem? Inbound
calls to my DID's are working fine.
Clearly,
I, on the other hand, have been disappointed repeatedly by their failures to
route international calls.
I've received e-mails from them promising a refund. I expect them to keep
their word.
On 3/24/07, Martin Joseph [EMAIL PROTECTED] wrote:
On 2007-03-23 14:37:18 -0700, Tom Lynn [EMAIL
Now I know where they've been spending my remaining balance...
On 3/21/07, Ira [EMAIL PROTECTED] wrote:
At 09:08 AM 3/21/2007, you wrote:
Does anybody know Jed Stafford? As far as I can tell this ended up
being a one-man or two-man operation. It's just sad.
I got a marketing email from
This is probably why they don't use PayPal anymore. Now, there is no
resolution process that I can pursue, other than complaining to the Gov't.,
which I have.
On 3/20/07, Vicky [EMAIL PROTECTED] wrote:
I got money back around 6 months ago . It was a via paypal claim and hey
didn't reply till
Has anyone been successful in getting a refund from SellVoip when you've
cancelled service?
Tom Lynn
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At this point, I'm simply contacting the State of Washington Attorney
General's office. They're ignoring my e-mails and I'm done monkeying
around.
On 3/16/07, Ira [EMAIL PROTECTED] wrote:
At 11:32 AM 3/16/2007, you wrote:
Has anyone been successful in getting a refund from SellVoip when
Do they appear to have failed as a result of Daylight Savings time?
On 3/14/07, Matt Putnam [EMAIL PROTECTED] wrote:
I didnt have them on tftp files they were all manualy configured. They are
not trying to request anything they have the tftp server address but are not
requesting any files. It
International calls (Germany) haven't completed since around 3/1. Domestic
works. Is it just me? I'm getting 503 responses.
Tom
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The question sounds like troll bait to me.
On 2/11/07, Edward Halman [EMAIL PROTECTED] wrote:
In the beginning...
I tried both. I found Trixbox to be a very effective out-of-the-box
solution for those venturing into the world of Asterisk IP PBX without
wanting to learn about dial plans. I
Dovid, you're killing me. This after asking if we can't all just be nice to
each other.
On 1/1/07, Dovid B [EMAIL PROTECTED] wrote:
Adam and bill are both wrong. The world revolves around me. Geeez cant we
cut the crap (i.e. Happy new year is followed by a response that hey it
isnt the new
Andrew, in your experience, what has changed from version to version. I
work daily with Avaya gear and do regression testing on new releases before
they hit the public. In my experience, I can't recall anything changing
with h.323 trunking other than the maximum number of trunk members managed
Sounds like an EBay ad...
On 12/30/06, Josué Conti [EMAIL PROTECTED] wrote:
Always...
Desire that in the New Year that if you really initiate...
It hears the words that always it desired to hear. It pronounces the
phrases that one day it desired to repeat.
It feels the emotion that always
I agree, he sent me one off list, too - making all kinds of allegations of
my sexual preferences. I sent him a link to AA, DrPhil, National Institute
of Mental Health and suggested he get some help.
On 12/28/06, Alex Robar [EMAIL PROTECTED] wrote:
Of course everyone is allowed to use VoIP...
Here's what he sent me after I told him to shut the up. I kind of
wonder if he's just trying to generate traffic at certain sites and it's
going to generate ad revenue for him in some lame scheme. Oh well:
Can anybody point me to a vendor that can provide a toll free number that
can be used in India to reach the united states? Verizon Business is
telling me they can't get one.
Thanks - Tom
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And it seems likely to me that you'll be sued for libel.
On 12/24/06, Al Bochter [EMAIL PROTECTED] wrote:
So you would deal with a criminal ?
Bret McDanel was *Convicted Of Cybercrimes
*
Best regards,
Al Bochter
Bochter Services
http://www.BochterServices.com/?t=Email
(VoIP PBX)
I second that. I'm quite happy with the IPKall.com did number I use today.
Only once in the last year was it unavailable when I needed it. So, not
bulletproof, but good enough for me to use all day when I work at home.
On 12/21/06, www.IPKall.com [EMAIL PROTECTED] wrote:
One way audio is
You're trying to teach a pig to sing. The uniden items you refer to
probably have their own internal answering machine, mine does. It's
designed to light the lamp only when it's own machine has a message.
On 12/8/06, Doug Crompton [EMAIL PROTECTED] wrote:
Thanks, but unfortunately that is an
It may not be what you're thinking, but I use Astlinux on an older PIII.
With a couple of options it has become my home router and works very well.
On 12/7/06, Dovid B [EMAIL PROTECTED] wrote:
Hi list,
Can anyone who has successfully ran asterisk on a home router please give
me the modell
How many channels do you require? I'd favor T1 for a few reasons. Higher
port density means fewer cards per system, which will mean fewer
interrupts. T1s won't require you to tune analog levels. Echo probability
will be lower.
On 11/29/06, asterisk-robert [EMAIL PROTECTED] wrote:
We are
Can you tunnel through a VPN connection?
On 11/28/06, Patrick [EMAIL PROTECTED] wrote:
On Tue, 2006-11-28 at 22:19 -0500, [EMAIL PROTECTED] wrote:
We have many clients who live in third world countries where the ISPs
purposely block traffic on port 5060.
I know we could always change the
Cepstral sounds good and it's cheap. However, it still sounds like a
synthesized voice.
On 11/28/06, Hall, Eric M. [EMAIL PROTECTED] wrote:
I'm looking to set up asterisk to call customer 3 days before the app and
remind them we will be out to see them.
I'm looking for any ideas on good
Earle,
I'm running Astlinux on a PIII 550 with 384 megs of ram. Booting from a
Compact Flash card. Non-Volatile storage on a USB Keydisk. I have three
SIP DID numbers in three different area codes here in Western Washington via
IPKall. I use a local SIP termination provider and also retain my
Vincent,
I do something similar to what you're doing. However, I use the CID number
as the astdb family, allowing me to assign multiple attributes as the keys.
It requires some maintenance, so I also wrote a php script for the
management. You can find it here:
quicktime player does it without adding any codecs.
On 11/24/06, Tim Panton [EMAIL PROTECTED] wrote:
On 21 Nov 2006, at 14:34, Jay Moore wrote:
Tim Panton wrote:
On 20 Nov 2006, at 21:46, Jay Moore wrote:
Doug wrote:
Hmmm. I think this may work for WinAmp and
incidently for Windows
I like a challenge. I'll let you know if I come up with anything.
On 11/26/06, Vincent Delporte [EMAIL PROTECTED] wrote:
At 12:00 25/11/2006 -0700, Tom Lynn [EMAIL PROTECTED] wrote:
By inverting the relationship, I found it easier to focus on the source
of
the call and the treatments I want
Jason,If you must stick with analog phones, you can find higher density channel banks that will host 8, 16 or up to 24 ports each. They communicate back to your asterisk server via your LAN. Or, as has been stated, you can purchase IP phones that also communicate back to your asterisk server via
Steve poses some good questions. In addition, I'd wonder how your trunk group in the definity is configured? Are you sending calling party name and number? If so, is your DS1 card set for protocol A,B,C, or D? For both calling name and number, I believe you need B, but I don't have my docs here.
Regardless, they're still perpetually lagged. I'm suspicious as to why paypal is conducting a review. For now, considering the poor performance, I stand by my decision to shop the market.
On 11/11/06, Vicky [EMAIL PROTECTED] wrote:
I doubt how many days more voxee will survive . Its been a month
Then I guess I'd better hurry up and use my remaining 49 cents worth of credit!!On 11/11/06, Vicky [EMAIL PROTECTED]
wrote:I doubt how many days more voxee will survive . Its been a month nw and tehir support doesnt want to repair this .
Ron,The guy is trying to help you. Go to the link and read it. There is a feature that you can use to play a recording into the voice channel. Mine is set so when you press #9, the caller hears the lots of monkeys recording.
The best part of it is that you can hang up and the recording will
Add me to the list. Not only lagged, but also failures to register. AND, apparantly Paypal won't automatically authorize payments to them anymore. I'm not recharging my account anymore.
On 11/10/06, Tim Panton [EMAIL PROTECTED] wrote:
On 10 Nov 2006, at 21:51, Rajeev Natarajan wrote: Same here -
Without providing a link to the list, or citing your front-runners, you can't really expect people to reply, can you?On 10/27/06, Frédéric Blaise
[EMAIL PROTECTED] wrote:Hello allI would like to know your opinions on free GUI used to manage Asterisk.
Which is better?My setup is quite small, about
astlinuxOn 10/25/06, Prasad Kandikonda [EMAIL PROTECTED] wrote:
We are looking at porting asterisk onto a embedded platform based on IXP or ARM. I would like to know the experiences of anybody who has already ported to these platforms. I am also particularly interested in issues related to
Are your sip phones capable of auto-answer?I can imagine you can terminate the incoming call into a meet-me conference (no pass code) and then trigger a script that creates a call file for each of the other participating phones. The auto-answer part seems like the sticky part.
On 10/15/06, Marc
I'm looking for an external device that can flash when there is new voicemail in a mailbox. I'm using an SPA3000 with a Uniden 5.8 ghz wireless phone system. Problem is, the Uniden system has it's own answering machine, which I don't want to use. But the message lamps are driven solely by the
its own.I gave up on the SPA 3000 due to echo problems.
http://www.amazon.com/Uniden-CLX465-Expandable-Cordless-Handset/dp/B0007TIYCUhttp://www0.epinions.com/content_70491344516Hope that helps, a little.
Bob...Tom Lynn wrote: I'm looking for an external device that can flash when there is new
I can get stutter dialtone using my spa3000, but the uniden doesn't respond to it by lighting the lamp. All it sees is an incoming call from the spa.It looks to me that I'll either need an external MWI device or I'm going to have to replace the Uniden phones.
On 10/14/06, Tom Lynn [EMAIL PROTECTED
FWIW, I too started with AAH, but got really upset when tempted with an upgrade and learning the path was a total re-install. I hear things have gotten better since.In response, I went completely minimalist and turned to AstLinux. My primary reason was my only hardware resource was a PC without a
I've never ordered from Voipsupply, but did forward two questions to their President. The first one was trivial and off-topic, yet answered very quickly. The second was product related and answered equally quickly and knowledgeably.
I'd definitely consider purchasing from them, based on my
I'm told the 4.0 release of communications manager will support up to 100 SIP phones without the need for an extra server. Are you trying to connect stations or trunks?On 10/13/06,
Andrey Kovalenko [EMAIL PROTECTED] wrote:
Hi everyone,I was wondering if anyone on this group has successfully
Dave,
Are you in the US?
On 10/12/06, Dave Cotton [EMAIL PROTECTED] wrote:
On Thu, 2006-10-12 at 21:30 +0200, Csibra Gergo wrote: Thursday, October 12, 2006, 6:58:57 PM, Tim wrote:
I've read alot of comments on the SPA-3000, many if not all saying they had echo issues, but I've not seen anyone
I'm keeping my Qwest line for this purpose.On 9/23/06, Christopher Corn [EMAIL PROTECTED] wrote:
Im using voipestreet and voxee for my SIP termination. neither of them, offer any kind of e911 service. as i search the web i see different companies that offer this e911 service to voip suppliers. I
That's simply the remaining rationalization that is left in the absence of the bridged line appearances.
On 7/25/06, Matthew Warren [EMAIL PROTECTED] wrote:
Subject: [asterisk-users] Re: Operator Console(s)/Shared Call Appearances Hi Folks, We're migrating from a conventional KSU/PBX to Asterisk
perhaps not what you're looking for, but reading thru your config, it looks like you've mis-spelled 'echo cancel' as 'echo cancle'On 7/23/06, Frank Darner
[EMAIL PROTECTED] wrote:
What is the output from 'cat /proc/zaptel/*' After delete of all Asterisk files and complete new install I got
Rich,I had the same problem and the solution was to take out a 'malformed' callerid value from my sip.conf entry.TomOn 7/20/06, Rich Adamson
[EMAIL PROTECTED] wrote:Tried the syslog debug, but it reports the exact same thing as the sip
debug shown below. It includes the INVITE, 100 TRYING, AND
I'm trying to debug why the message waiting lamp on my phone won't light. I suspect it doesn't adhere to standards.I used tcpdump to capture it's bootup sequence. From the dump, I can see the phone trying to subscribe to my asterisk server (
1.2.7), to which it receives an initial 401 unauthorized
You can place the phones at each house in a different context. Trunks, too.
On 7/10/06, Andrew Niemantsverdriet [EMAIL PROTECTED] wrote:
I have a asterisk box up and running great. I have another house in mybackyard that also wants to use my asterisk box. I am running trixbox
now and have two
Doug,
Cheer up! There's some great beer brewed in Montana! Have a Moose Drool and get down to some creative resume re-inventing.
On 7/6/06, Kevin Savoy [EMAIL PROTECTED] wrote:
I'm in Williston, North Dakota and we have an office in Billings, MT. He'sright. We are 500 miles form civilization! :)
phones to work.c
However I'm a bit disappointed to leave things as they are, I have a feeling of ... failure? I guess I'll still try some thing or another in my (inexistent) spare time.
Thanks for your help,
Silviu
From: [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED]] On Behalf Of Tom Lynn
Sent: 04
I don't know of anybody using it inbusiness, but I'm curious to find out if there are any user groups formed or forming in the Seattle Area.
On 7/5/06, Douglas Garstang [EMAIL PROTECTED] wrote:
All,Anyone know of any companies (small, large) that are using, experimenting with, deploying, and so
Doug,Two Points- User groups are excellent places to network and make contacts- Asterisk skills will translate into other telecom or non telecom related fields. I work for a fortune 500 company and I'm responsible for a bunch of Avaya systems. None of the people on my team had Avaya experience
Herchi,I want you to re-read my last e-mail very carefully. Your response does not mention at all my guess that the three SP_DIRSRVR variables may be giving you trouble. I'm still interested in knowing what happens if you remove them from your settings file.
Also, I have heard a rumour that there
Is the text shown below the ENTIRE file? It looks like all of the settings for the individial phone models are missing. I'm not sure what the consequences of branching to the 4610 section will be if it doesn't exist. Also, I don't use the SP_DIRSRVR values. What happens if those three entries are
I'm setting both values like you are:
SET SIPREGISTRAR xxx.xxx.xxx.xxx
SET SIPPROXYSRVR xxx.xxx.xxx.xxx
I don't notice a difference in how these settings appear in our respective 46xxsettings.txt files.
On 6/29/06, Henk [EMAIL PROTECTED] wrote:
Did you try to manually to change the
I too am using 2.2.2, but I'm loading my config files via HTTP. I was having some difficulty when I was using TFTP. Things were not as reliable for me, so I switched to HTTP. I've been stable since.
On 6/28/06, Herchi Silviu [EMAIL PROTECTED] wrote:
Hi Tom,
Thank you for your interest in my
Which version of firmware are you using?On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote:
Hi all,
I've been pulling my hair out for two days over this problem… I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem!
I have two Avaya
The avaya softphone (an entitlement in recent versions of the PBX) will dial from outlook and allows clicking of phone numbers from within web based content. I'm not sure it works from other office apps, though.
On 6/27/06, Rodney G. McDuff [EMAIL PROTECTED] wrote:
Brian Capouch wrote: It will be
Those are the only files that come to mind. On 6/23/06, Erick Perez [EMAIL PROTECTED] wrote:
Tom, just to make sure im on the right track.What files do you tweak?sip.conf, the ones from avaya and anything else?On 6/22/06, Tom Lynn [EMAIL PROTECTED] wrote:
Nope.Let me know if you do.I've suspended
. Besides that, the
phone is great and the audio quality is superb.Did you managed somehow to make the MWI work?Will keep searching the net, the 4602 page is somehow poor on the documentation.On 6/21/06, Tom Lynn
[EMAIL PROTECTED] wrote: Well, I wouldn't say nobody.I do and I've corresponded with a few
Well, I wouldn't say nobody. I do and I've corresponded with a few people that do. There's a page on voip-info.org dedicated to the Avaya 4602 telephone and SIP (I'm hoping I'm not the only reader of that page). When I've used my Avaya phone in conference (FWD CoffeeHouse), I've had people
HDSL can sometimes deliver service where copper pairs are nearly exhausted. In other words, if you're down to your last pair of copper, a normal two-pair T1 cannot be delivered, whereas T1 via HDSL can.
On 6/17/06, Jean-Michel Hiver [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] a écrit : Thanks for
Whare are they located?On 6/17/06, Bob Knight [EMAIL PROTECTED] wrote:
I have 4 sparc based sun boxes I am about to pay money so I canget rid of them.They are running older versions of Solaris.You should be able to load Solaris 10 and play around with *on them.Time to clean the office:
3 Ultra 51
Don't forget to be sure your power supplies are reliable, and if necessary redundant.
On 6/13/06, Colin Anderson [EMAIL PROTECTED] wrote:
Maybe you are over thinking it. I have 200 users on a quad Xeon 700 with 2PRI's and we regularly have 40-60 channels up, no problem (believe me, if
there was a
I'm trying to get to where I can program a phone to have 3-6 buttons each representing the same extension number. Also, I'd like to have them appear on more than one phone like key systems do.Is asterisk able to set up shared lines in this manner yet?
Tom
The 4606 is a h.323 based phone. There is no SIP image to use with this phone.
On Fri, 12 May 2006 11:11:48 -0500, you wrote:
Hello all,
I have asterisk working well with, Sipura, but I do not manage to form
several phones avaya 4606, someone could have formed one avaya with
asterisk?
is it
On Tue, 15 Nov 2005 11:51:33 -0500, you wrote:
On 11/15/05, Brian Roy [EMAIL PROTECTED] wrote:
On 11/14/05, BJ Weschke [EMAIL PROTECTED] wrote:
There is a known issue right now where using mixmonitor with
chan_local is going to cause an unintentional disconnect. Are you
using Local/ with
Faris,
Is there a way to have * send save these in an off-server location? Or
have * e-mail them via smtp and then delete them from the server
automatically?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Raouf
Sent: Saturday, December 24, 2005
.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Raouf
Sent: Saturday, December 24, 2005 10:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] recording queue calls
Tom Lynn wrote:
Faris,
Is there a way
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
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12/16/2005
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To
I've always received a busy signal when I dial mine. Additionally, I
can't see any server messages to indicate that goiax is even
attempting to call my system, although I continue to trouble-shoot.
I have a 413-230- number.
Thanks
On Thu, 20 Oct 2005 15:32:08 -0600, you wrote:
I've been
Seems to, mostly, but not all the time.. There have been some
previous posts about this phone. I'm waiting for the next firmware
release before testing further.
Mute kills audio in both directions. Mine doesn't seem to want to
dial any numbers that have * in them like *98.
My Sipura 3000
On Sun, 9 Oct 2005 14:28:14 -0600, you wrote:
The initial release of Avaya's SIP firmware for the 4620 phone was
released on August 17, 2005. It is available from support.avaya.com.
That said, I have had mixed results with this phone. I'm sure it
works with Avaya gear, but it's glitchy with *.
On Sun, 02 Oct 2005 00:53:03 -0700, you wrote:
I wrote a very very simple shell script and an even simplier macro to
use the IBM TTS engine within asterisk for prompts. While its free you
are limited on the number of requests you can do within a day.
If anyone is interested its available at
I've just interfaced an Avaya 4621 set to [EMAIL PROTECTED] It's running the
2.2 SIP firmware released August 17th.
I've run into some strange behavior with this phone. 1st, in order to
get two way audio, I had to tell * that it was behind a NAT even
though it is on the same subnet as the *
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