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
Are the Asterisk Implementation below recommended?
Scenario One:
Asterisk PBX is installed as gateway with two NICs. The internal NIC
servers a network LAN of only Polycom IP phones while the other NIC is
the public interface(static)
I also have a few Polycom IP phones connecting from homes.
Thermal Wetland ha scritto:
Does anyone know if you can have multiple TE110P cards in one chassis?
One server with two TE110P, shared interrupts, APIC routed irq, all
things near ok. Sometime only one of these run HDLC error and some
strange error, i think a 0,05% probability of error.
:-)
* Phones = stations, regardless of where they areAsterisk = SIP Server, Phone = SIP Client
Is a Media Server a Phone (ie SIP Client) ?
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On Fri, Oct 13, 2006 at 07:00:54PM -0500, Eric ManxPower Wieling wrote:
* Phones = stations, regardless of where they are
Asterisk = SIP Server, Phone = SIP Client
* Trunks = trunks to other SIP servers, bilateral
Asterisk and the other server is peer to peer
* Services = services you
But, the CDR record looks strange (and this is the only common pointbetween those calls): Both the session timer and the talk timer are the
same, but according to the log, the call are all answered after 3 to 5seconds ringing (so those timers should show this difference).Could you elaborate ?Do
On Sat, Oct 14, 2006 at 09:34:33AM +1000, Paul Hales wrote:
The Zulty's 4x5 does (or did) fwiw.
Thanks. voipon.co.uk has them for GBP 299 or $557.64 (gulp)
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I am experiencing the same issue. However, I have not tried the
VersionStamp field and will do so tomorrow.
If you find an answer please post it to the list.
On 10/13/06, Tim Connolly [EMAIL PROTECTED] wrote:
Does anyone know what triggers the 7970 to update its config? I
was able to
I'm now running kernel-2.6.9-42.0.3.EL
Not really an answer to your question, but I found out all kernels above
2.6.16 do a better job on asterisk systems then the ones before that. No idea
how this is possible as I'm in no way familiar with the inner workings of the
linux kernel, but I
From: Remco Barendse
Possibly, but I would have to start worrying about
kernel configs, compiling the lot and solving the
problem of the box no longer being able to boot the kernel :)
You'd be better off starting with a Fedora kernel. Unfortunately RHEL/CentOS 4
is based on Fedora Core 3
Dear folks,
I couldnt receive faxes and get the following debug traces on the console, I
appreciate any help or even hints. Using Spandsp-0.2
app_rxfax.c:76 span_message: FLOW Get at 9600bps, modem 1
app_rxfax.c:76 span_message: FLOW Changed from phase 3 to 5
app_rxfax.c:76 span_message: FLOW
Hi,
I am configuring an audiocodes Medant1000 to talk to my asterisk box.
So far I have successfull in landing a single call from mediant to my
*box. my sip conf is as follows:
[general]
context=sip
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[3911700]
type=friend
host=dynamic
dtmfmode=info
Hi,
http://www.openvox.com.cn/products_detail.php?genre_id=17id=45
The A1200P is a 12 port card, that used the same modules as a TDM400P.
I have been looking at this card, and I want to know if anybody has used
this card and what their experiences were?
thanks,
yusuf
--
This message has
Tom,
The uniden TRU446 and the CLX465 both are supposed to detect stutter
dial tone (SDT) from the phone company and light the MWI. When used
with asterisk the SPA3000 can generate SDT. I'm not sure it can do so
on its own. I gave up on the SPA 3000 due to echo problems.
Contact them again... they have always been very good... I'm chocking
this up to the snow storm.
On 10/13/06, Shaw Terwilliger [EMAIL PROTECTED] wrote:
Matt wrote:
Hi,
Does anyone know what is going on with voipsupply? My sales guy
hasn't been online in several days, their 800 number is
My uniden phone is the TRU8885-3HS.On 10/14/06, Bob Chiodini [EMAIL PROTECTED] wrote:
Tom,The uniden TRU446 and the CLX465 both are supposed to detect stutterdial tone (SDT) from the phone company and light the MWI.When used
with asterisk the SPA3000 can generate SDT.I'm not sure it can do soon
Hi,
I like Centos as a basic platform, but I always then upgrade the kernel to
the latest stable release from kernel.org
The latest ones are using Centos 4.4 x86_64 with kernel 2.6.18
For simplicity, I always start with the .config from the original Centos
kernel.
My install sequence is:
Untar
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
Steve,
Is RAND available in the latest trunk or do I need the 1.4 beta?
If I do show function RAND it says its not available.
Thanks,
Jon
- Original Message -
From: Steve Murphy [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, October 14, 2006 12:30 AM
Subject:
Tried them for all three - a tad pricey but good service imho.On 10/12/06, Crazy Boy [EMAIL PROTECTED]
wrote:Hi,I want to register with
http://www.inphonex.com VoIP provider. I want to configure my Trixbox and Asterisk servers with inphonex. Anybody using this service? Mainly, I want to do three
Use the AGI I sent. It looks like the email did not put a CR
correctly.
Run it from the commandline and see if you get output.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jon Weisman
Sent: Saturday, October 14, 2006 12:45 PM
To:
On 2006-10-10 23:14:45 -0700, Martin Joseph [EMAIL PROTECTED] said:
On 2006-10-10 20:25:44 -0700, Nic Bellamy [EMAIL PROTECTED] said :
I am seeing occasional stuck SIP channels that seem to occur when the
fricking Nokia E60 drifts out of WIFI range in the midst of a call.
snipI wonder if
try http://astguiclient.sourceforge.netOn 10/7/06, Marnus van Niekerk
[EMAIL PROTECTED] wrote:
Yes, you can easily use asterisk for a call center, start looking here
http://www.voip-info.org/wiki/view/Asterisk+call+queues
M
Imed Imed wrote:
Hi,
I'm a novice in
yes to ztdummy: but you may have trouble when you try and run multiple simultaneous meetme sessions.On 10/5/06, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:omar parihuana wrote:
Is possible use meetme feature without Zaptel card? (ztdummy will be the solution? )Yup. :P Thanks in
On 2006-10-11 03:22:00 -0700, Thomas Kenyon [EMAIL PROTECTED] said:
I have been seeing this problem for a long time and it occurs in
1.4.0b2 (as well as 1.2.0-1.2.12.1).
If the internet connection is lost and I have SIP services that require
me to register, any SIP devices attached to the
On 2006-08-21 02:44:55 -0700, Benny Amorsen [EMAIL PROTECTED] said:
MR == Matt Riddell (NZ) [EMAIL PROTECTED] writes:
MR And so you're thinking it would be better to run several hundred
MR Asterisk instances?!
Why not? As long as you stay away from the things that need zap
timing, asterisk
On 2006-10-14 07:36:51 -0700, Matt [EMAIL PROTECTED] said:
Contact them again... they have always been very good... I'm chocking
this up to the snow storm.
Yes, might still be too early, I see over 200K still without power in
there neck of the woods (Buffalo, NY).
Massive tree damage
Sorry, just checking if my mail is working.
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MJ == Martin Joseph [EMAIL PROTECTED] writes:
MJ I added the rtptimeout=60 to my general section in sip.conf, and
MJ now when the e60 goes out of wifi range, 61 seconds later, my
MJ channels are clear! Sweet.
Does this work with canreinvite=yes? (I can't see how it could, but
I'd like to be
Tom,
There are a couple of SIP based cordless phones out there. A little
pricey, however. Such as:
http://www.voipsupply.com/product_info.php?manufacturers_id=35products_id=923
It might be compatible with your existing cordless hand sets. Uniden
seems pretty good about that.
Or:
What's in zapata.conf?
On 10/13/06, Remi Quezada [EMAIL PROTECTED] wrote:
When I reload the asterisk I get the following warnings:
Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring
switchtype
Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring
signalling
Oct
Doug Lytle wrote:
Another question,
Is anybody using the Tellabs 2572 EC with a PRI. When we moved from
our analog lines to a PRI, I thought it would be simple stuff moving
the EC to the PRI. Changed signaling, made sure that channel 24 wasn't
being ECd and everything came up. But, I was
-- Forwarded message --From:Alexander Lopez
[EMAIL PROTECTED]To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate:Sat, 14 Oct 2006 13:04:08 -0400
Subject:RE: [asterisk-users] Re: Generate Random Numbers in dialplanUse the AGI I sent. It
On Fri, Oct 13, 2006 at 09:05:54PM +0100, David Bath wrote:
Hey all,
Just as an update, incoming calls are fine. I have had several long
calls today inbound on the PSTN with no drops.
From the log it does sort of look like a hangup is being detected.. but
its certainly not correct!
According to docs I've seen this is the same codec that they use in the "Communicator" product that they target at Skype users. I have one of these and it does sound great. It's integration with the Skype API is primarily for the buttons that access the Skype app to start/stop calls. Aotherwise
I first learned asterisk via [EMAIL PROTECTED]
Then I went to straight asterisk.
This seems to be a theme. Getting your feet wet with [EMAIL PROTECTED]/Trixbox
is not
a bad way to go, especially if you want to get a functioning system up
and running quickly. After tinkering with
Hi there,
Thanks for the reply. I pasted everything from the console, with logger
set to all, and verbose set to around 15. (as asked in original email)
I have the whole session, from placing the call to the call hanging up,
but it's pretty long, so I wasn't sure if you wanted to full thing.
Remco Barendse [EMAIL PROTECTED] wrote:
I'm not running trixbox but normal Centos 4 with asterisk installed. I
tried to find some further info on this but couldn't find any.
Do audio problems occur with normal Centos and the latest kernel version
too? (In other words, should every centos user
On Sat, 2006-10-14 at 12:00 -0700,
[EMAIL PROTECTED] wrote:
Steve,
Is RAND available in the latest trunk or do I need the 1.4
beta?
If I do show function RAND it says its not available.
Thanks,
Jon
Jon--
Forgive
I am currently doing my thesis on an implementation of Video
into Asterisk using H323
So I know that they are various mailing lists that demonstrate that SIP is the
way forward, but sometimes
It helps to use old equipment that one already owns
so I am just looking for some simple ideas
Well, this is mostly just new. I thought this would be a good place to announce the opening of the forums on The Asterisk Blog. Check it out and leave a few messages to help get it's feet wet. Thanks guys!
http://www.AsteriskBlog.com/forum-- www.AsteriskBlog.comYour home for easy to learn Asterisk
Hi,
I've finally given up on trying to fax over my Digium TDM400 card.
I've found that fax over VoIP is quite more reliable (at least I can
receive the faxes).
My ITSP supports G729 and alaw/ulaw. As I won't be receiving faxes
everyday (just ocasionally), i pretend on using g729, unless a fax
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
Hi,
I am looking to replace a quirk of our old PBX system functionality with
asterisk but after searching, archives, wiki, etc.. I cannot figure out
how.
Here is what I would like to do:
PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a
SIP ATA. When an incoming call comes
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