Hi,
> -Original Message-
> Actually an eight slot wire-wrap chassis came with the 2531's
> and they're pin-compatible with the 257x cards. I took it
> home and went to town on it with an angle grinder and tig
> welder and, voila, four two-slot chassis. As for power
> supplies, eBay pr
Hi,
> -Original Message-
> You should be able to download one (for WIndows and possibly Mac) from
> efax or j2.com I think.
>
> http://www.efax.com/en/efax/twa/page/download?rqcp=2
>
> http://www.j2.com/jconnect/twa/page/download
You might be able to do that, but take a good look at th
Good day all
Why do I need a Zaptel card to do trunking in IAX??
What if I only had a "voice/iax" router?
Is there a way around this?
Thanks
Altus
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Richard Cook wrote:
We use PostgreSQL in house. It performs wonderfully and cross-platform
drivers (ODBC, .NET) are way further along than MySQL. We switched from
MySQL a couple of months ago and have never been happier.
We use Postgres exclusively too (12 databses, several of them with
several
Hello to all,
I'm trying to build a PBX using Asterisk. I have a single BRI ISDN
line and I need to connect 4 internal normal phones and a couple of
softphones on PC. I have bought a single port Billion S0 card and a
TDM400 with 4 FXS modules for the intenal phones.
ISDN lines here in Greece termi
I just upgraded 4 boxes to 1.0.6 without issue, then I went to upgrade
my personal test box which I am playing with call logging cdr stuff on,
writing to postgres and now asterisk now crashes on boot with the
following error.
[app_while.so]Mar 10 17:20:04 WARNING[7239]: loader.c:258
ast_load_re
Jim Van Meggelen wrote:
[EMAIL PROTECTED] wrote:
I need to setup all area codes for billing, but how can I do that easy
for North America and Canada, where I have only one price anyway.
Country code 1, and just exclude any Carribean nations that you need to
handle differently.
while it m
Say Jim, that bridge in Manhattan I bough from you last yera? It's
still rusty! :)
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http://li
> My problem is as well described on
> http://www.voip-info.org/wiki-Asterisk+i+extension
Yup. I believe the 'i' reacts to dialed digits input during that
particular extension. There aren't any. That wiki page concludes that
you need a "fallthrough" extension like
_.,1,Playback(Idiot)
_.,2,Hangup
Hi,
my 7960&7914 is working fine with asterisk 1.0.6 and chan_sccp
regards
Jens
Quoting Remco Barende <[EMAIL PROTECTED]>:
Am I the only one seeing problems with chan_sccp and the latest Asterisk
stable? Is there anyone where it is still working?
My phones disappear after half an hour and are seen
>
> I know this is a bit off topic but we are using Asterisk :)
>
> Since this list is full of tech gurus w/ all different sorts
> of backgrounds, I thought I would get the best opinions here.
>
> We have several different switches and other telecom
> equipment at our facilities which all have
Hmm..
Seems a factory reset from within the web interface was all thats
reuqired to make it pay attention to the settings.
After a factory reset, then reboot and configure in the subscribe url
for the config etc, it all comes to life and plays
nice. Perhaps this is a bug that snom needs to addre
> This works exactly as i want so users basically can dial 0, wait for the
> dialtone and then dial the requested number.
>
> The only problem that i have is that from when a user dial 0 to when i get
> the dialtone from the telephone line, something like 5 seconds pass... is it
> possible to pull
On Thu, 2005-03-10 at 00:37 -0600, Jay Milk wrote:
> So much for the history. As for usability, IB is your typical (almost)
> ANSI SQL-92 compliant database engine. It supports RI, triggers, stored
> procs, just like we all like'em. Its engine is touted for the
> "superserver architecture" but i
Hello
Well i think that overlapdial=yes would be required if i am trying to dial from the asterisk side, whereas in my case i am trying to do the opposite.
I think that asterisk would enter the overlap receiving if i send it a setup request with either no called number or incomplete called numbe
Hi,
I have this scenario.
UA1 <==> AS <> UA2
UA1 : User Agent 1
UA2 : User Agent 2
AS : Asterisk
AS has been configured with UA1 & UA2 users.
Registrations are happening correctly. But..
UA1 <> AS <==> UA2
SUBSCRIBE to UA2 --->
<-
> Does anyone know of a Print-to-Fax client that works with asterisk &
> spandsp? Astfax is a partial solution but that only lets us email the fax
> in, we'ld like to set it up so the user can hit the print button and send
> the fax (even if all it does is email - transparently to the user - the fa
Any suggestions how can I get asterisk to play
MOH (music on hold) a MP3 radio stream from the internet (http:// location)
instead of a MP3 file in the mphmp3 folder?
I tried putting default => quietmp3:http://www.waixwave.com/pacnet.pls
instead of default => quietmp3:/var/lib/asterisk/mohm
Firebird started at Interbase. Around version 6.0.1 Interbase "went
open-source" and published all available source-code for Interbase,
quite possibly in an attempt to enter formerly closed markets. Since
then, IB was further developed (closed-source) into version 7.0 and now
7.5, and Firebird re
[EMAIL PROTECTED] wrote:
> I need to setup all area codes for billing, but how can I do that easy
> for North America and Canada, where I have only one price anyway.
Country code 1, and just exclude any Carribean nations that you need to
handle differently.
It'll be a much shorter list than tryin
[EMAIL PROTECTED] wrote:
> Guys.
>
> Anybody has a URL or some document with comparison charts
> with Asterisk's features against other PBXs?
I would argue that what you ask is in some ways impossible. Asterisk is
orders of magnitude more flexible than any other PBX you may have
encountered, beca
Hi Tim,
Thanks for your help.
I've already tried copying and pasting the text from the settings page
within the Snom 190's web configuration
page. I've also tried putting it in without the < or > symbols.
I've also tried just putting in:
dest sip:[EMAIL PROTECTED]
dest [EMAIL PROTECTED]
dest 1000
Well, I'm using intercom with a couple SPA-841. Currently using * CVS from
a couple weeks ago and a macro in extensions.conf. It should give you some
idea:
[macro-intercom]
; ${ARG1} - Extension
exten => s,1,SetCDRUserField(intercom)
exten => s,2,SIPaddheader(Call-Info: \;answer-after=0)
e
I need to setup all area codes for billing, but how can I do that easy
for North America and Canada, where I have only one price anyway.
thanks!
bye
Ronald
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I want to implement a one way announcement and paging facility using
Asterisk and Sipura phones. The wiki says Sipura phones only support
Auto Answer using the Call-Info header which is no lone shipped with
asterisk stable since 1.0.4.
I would like to ask if anyone has implemented a s
> > Has anyone got this to work? Under Idle Display Animation, the
> > administrators guide says "For example, a company logo could be
> > displayed"..
>
> > In the ipmid.cfg file, I enabled 'ind.idleDisplay.enabled'
> (ie changed
> > it to 1), and under the IP 500 section, I added an entry
That was cool... but I don't think it helps other for in fixing their *
problems!!!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas Roedl
Sent: Thursday, March 10, 2005 11:40 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Voicemai
>From macintouch.com:
Apple is distributing an open-source Asterisk install package for Mac OS X:
A complete IP-PBX in software. It runs on Linux, FreeBSD, MacOS X and
Solaris and provides all of the features you would expect from a PBX and
more. Asterisk does voice over IP supporting all major V
Eric_Doiron wrote:
Try specifying the contexts ... just an idea.
exten => 1,4,VoiceMail([EMAIL PROTECTED]&[EMAIL PROTECTED])
Remember these are voicemail.conf contexts, not extensions.conf contexts.
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Hi;
Well for the firmware go to:
http://www.snom.com/download/share/
To see how it wants the settings, manually configure a phone and look
what the Settings
tab at the bottom of the bottom of the Lefthand side menu shows...
Later;
Tim
On Mar 9, 2005, at 9:35 PM, Shaun Dwyer wrote:
Hi All,
I reali
I just found out about Asterisk (yes, from the Slashdot posting), and I
would like to set up my old computer as a dedicated box for my house using
[EMAIL PROTECTED] However, when I try to install from the bootable CD, it gets
to 54% of copying the image to the hard drive and then says that an erro
There is a mention that the current Sangoma T1 cards (A10[1,2,4]) work
with * using their WANPIPE drivers. Has anyone used any older Sangoma
cards that also support WANPIPE ?
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Hello
I setup call parking using asterisk, it works fine for the Ist time, I am able
to park the call and i setup parkingtime => 120. After 120 sec. when call came
back to receptionist, she is not able to retransfer the call. Press # is of no
help. Is it a standard feature or I am doing something
Guys.
Anybody has a URL or some document with comparison charts with Asterisk's
features against other PBXs?
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Try specifying the contexts ... just an
idea…
exten =>
1,4,VoiceMail([EMAIL PROTECTED]&[EMAIL PROTECTED])
-E
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rick Harby
Sent: Wednesday, March 09, 2005
5:53 PM
To:
asterisk-users@lists.digium.com
Subjec
On Wed, 9 Mar 2005, Michael Graves wrote:
> On Wed, 09 Mar 2005 12:54:34 +0400, Jean-Michel Hiver wrote:
>
> >Leo Ann Boon wrote:
> >
> >>
> >>>
> >>> Another question... Are you aware of a SIP ATA or phone that has some
> >>> kind of VPN (i.e. PPTP) client embedded in? This would make the NAT
> >
the issue is lack of sidetone
u can google sidetone
sidetone is feedback u get from the mike to your earpiece that
the fone generates to let u know the circuit did not go dead
when people stop talking
i find the lace of sidetone extremely annoying and so will many customers
with asterisk
i have fou
Hi All,
I realise this is off topic, but its likely the best place to ask!
I sent an email to snom support a few days ago but have yet to recieve a
response..
Perhaps some one has found a solution to this problem already? I've searched
the mailing lists and google and found nothing useful. I've al
Excellent, thanks for the info. That hiss is interesting; maybe all the
extra wire makes it tougher for the SPA to drive. In any event it sounds
like it will do what I want. Thanks.. :)
-James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Luki
Sent:
Hello!
One of our guys created a short rap sequence (7 MB):
http://www.nativeinstruments.de/tmp/vmrap.wav
Andi
--
-> Andreas Roedl-> Senior IT Manager
-> NATIVE INSTRUMENTS GmbH -> [EMAIL PROTECTED]
-> Schlesische Strasse 28 -> http://www.native-instruments.de/
-> D-10997 Berli
did you try the auto-config?
--- Aaron Glenn <[EMAIL PROTECTED]> wrote:
> Feel free to hit me with a cluestick; maybe it will
> jar something lose...
>
> I'm running a stock [EMAIL PROTECTED] 0.6 install and after
> four days of
> excrutiatingly annoying debugging, I finally came
> across the obv
This couldn't be futher from the truth. I use IAX2 to connect to termination providers, and I have MOH working fine. You may need a timing reference like a zaptel card or software equivalent (ztdummy?) Check the wiki at www.voip-info.org
Michael
--Original Message Text---
From: [EMAIL PROT
On Wed, 09 Mar 2005 12:54:34 +0400, Jean-Michel Hiver wrote:
>Leo Ann Boon wrote:
>
>>
>>>
>>> Another question... Are you aware of a SIP ATA or phone that has some
>>> kind of VPN (i.e. PPTP) client embedded in? This would make the NAT
>>> problem go away nicely and provide added security...
>>
Original Message -
From: "dean collins" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
Sent: Friday, February 18, 2005 6:04 PM
Subject: [Asterisk-Users] [EMAIL PROTECTED] festival weather report
This script was developed by Mark Johnson.
All I did (
Yes, Ironic isn't it. That SIP/VOIP is so digitally clear that you don't
hear the pops, cracks and whistles of the old analog phones. The only
analog is from the human to the machine. The old analog phone humans
hear it, soon there will another generation of humans who have never
used an analog pho
Maybe a long shot but if you run asterisk as non root user have you checked
the permissions on /dev/capi20 ? I have an eicon 4BRI card and every time I
reconfigure the card with divas_cfg I do a chown --recursive
asterisk:asterisk /dev/capi2*
Craig
- Original Message -
From: "Junk Mail"
James,
the SPA's have a "FXS Port Power Limit" which is 3 by default. The
range is 1-8. I have a SPA 2000 with 6 phones on one port (entire
house wired to one port) which works OK as long as your phones do not
rely on power from the line (cordless phones, etc. are fine). On the
second port I have
On Wed, 2005-03-09 at 15:02 -0600, Steven Critchfield wrote:
> On Wed, 2005-03-09 at 15:43 -0500, [EMAIL PROTECTED] wrote:
> > For some reason I didn't think PostgreSQL was for mission critical apps. I
> > don't think I have any reasoning behind it, just didn't think it was
> > "hardcore"...soun
I concur. I rebuilt today and now I seem to be able to dial out.
MARK.
Chris Nibeck wrote:
thank you everyone!
It does not seen that it was configuration problems at all.
It appears it was the CVS that I was using from yesterday.
I decided to start over, downloaded the latest CVS, recompiled, and
Hello all..
I have what I think is a rather simple question here. We're using * for
home and business lines with great success so far. (inbound, outbound,
several hard phones, voicemail, IVR, etc)
My next project is to get some of my traditional POTS phones at home on the
system. I understa
I emailed ipVolution about it, and they don't have it out yet. Their
cards should be coming out soon, they say. Although the target release
month was last month, I believe.
--
Dana
On Wed, 09 Mar 2005 15:21:35 -0800, Chris A. Icide <[EMAIL PROTECTED]> wrote:
> If you do a web search for ipVolutio
I configured this once now I forgot what I did.
Two Broadvoice accounts.
Incoming is simple - just use the phone numbers.
Outgoing:
Dial out on a specific line
and/or
set up the groups and select the other "line" if the first one is busy?
--
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texar
Someone else mentioned this card earlier and it looks
promising for a way to get your 4 E1s up and have DSP on the
card.
If I were provisioning today I would consider this one for
sure though I am betting there is no real data available
yet.
ipVolution TDM120
http://voipstore.atacomm.com/s
Hi all,
Can someone point me to some information on the type of
hardware that might / should be used for a high load on an asterisk machine ?
I know that this is dependant on what services you plan to
have running, and it’s relevant to what you plan to do.
We are likely to be ru
buy either a Viking electronic module
(http://www.vikingelectronics.com/) that supports analog trunk
interface and plug the paging system into the viking module, or you
can buy the same thing from bogen (http://www.bogen.com/), they also
have voip gateways to do this. These companies specialize in
> Has anyone got this to work? Under Idle Display Animation, the
> administrators guide says "For example, a company logo could be
> displayed"..
> In the ipmid.cfg file, I enabled 'ind.idleDisplay.enabled' (ie changed
> it to 1), and under the IP 500 section, I added an entry for the bitmap
> t
If someone could explain me this (added some extra debugging code
Mar 10 00:47:13 NOTICE[7561]: chan_sip.c:8448 handle_request: Registration
from '' failed for '172.20.41.7'
Mar 10 00:47:13 WARNING[7561]: res_config_odbc.c:97 realtime_odbc: SQL param
name is a7
Mar 10 00:47:13 WARNING[7561]: res
If you do a web search for ipVolution TDM120, you should find someone who
claims to have a card that does such a thing.
-Chris
On 02:21 PM 3/9/2005, Brandon Patterson wrote:
>
>Supposed to be someone in Minn. working on this. I heard the name Dan.
>He might be in the hardware biz.
>
>
>> On Wed,
Hi all!
After much struggling I got my [EMAIL PROTECTED] working fine AND making use of
two
AVMFritz!PCI cards. Really nice ! (kernel 2.4.2x)
There's however a silly glitch that's getting on my nerves, and, kind of a
newbie that I am to linux, it should be easy to get help :
-- "capiinit start" M
Have a fair amount of asterisk experience, but this one is
blowing my mind.. I have a context setup as follows:
[department-listing]
exten =>
s,1,Background(custom/6000)
exten =>
s,2,DigitTimeout,5
exten =>
s,3,ResponseTimeout,30
;
exten =>
1,1,Answer
exten =>
1,2,Wait(1)
exten =
Hi to all,
I am sending a SIP REFER message to Asterisk from a VoiceXML application using
the element to do a Transfer through Asterisk.
I need to know if Asterisk supports the full features of the SIP REFER message
because if i set 'bridge=true'
in the element of the VoiceXML application to
On Thu, 2005-03-10 at 08:05, Martijn van Oosterhout wrote:
> Hi,
>
> I see applications for signalling busy, congested, ringing, progress
> etc, which I understand can be provided either in or out of band. But
> all I want to do is generate a dialtone. This obviously can only be
> done in band.
>
Why would you bother 2 x grandstream
handsets, 15 mins to open them up, cut the cable and solder a plug or connect
them to your amps and problem solved.
$40-$50 each channel – lol you can
have 10 zones J
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Je
Hi ALL,
Below is what I did :
Installed FC3 , Zaptel Drivers , libpri, Asterisk.
E1 Card is - TE110P
When I run modprobe it gives me the output below:
Mar 10 04:18:26 asterisk kernel: ACPI: PCI interrupt :00:0d.0[A]
-> GSI 11 (level, low) -> IRQ 11
Mar 10 04:18:26 asterisk kernel: Controll
Does anyone know if its possible to have more
than one sound card in an Asterisk box and use each one as a paging zone?
How about left and right channels of a single sound card? I'm looking to have 2
paging outputs if possible - I've read about using a Grandstream phone on
autoanswer but I'
Feel free to hit me with a cluestick; maybe it will jar something lose...
I'm running a stock [EMAIL PROTECTED] 0.6 install and after four days of
excrutiatingly annoying debugging, I finally came across the obvious
problem. The zaptel driver is reading /etc/zaptel.conf; which by
default configure
Supposed to be someone in Minn. working on this. I heard the name Dan.
He might be in the hardware biz.
On Wed, 2005-03-09 at 15:18 -0600, Matthew Boehm wrote:
Again, may be off topic but are there any cards out there supported by
asterisk that have on-board DSPs to do better 729->711 or 729->PRI
They must have. The mean streets of lists.digium.com get to me
sometimes. Wind up getting a little sensitive about the "RTFM" answers
that people throw out sometimes so I just barked a little early. Again,
my apologies.
Just one of those days
Regards,
Wiley
-Original Message-
Fro
On Wed, Mar 09, 2005 at 03:02:03PM -0600, Steven Critchfield said:
> On Wed, 2005-03-09 at 15:43 -0500, [EMAIL PROTECTED] wrote:
> > For some reason I didn't think PostgreSQL was for mission critical apps. I
> > don't think I have any reasoning behind it, just didn't think it was
> > "hardcore".
On Wed, 9 Mar 2005 14:48:50 -0700, Wiley Siler <[EMAIL PROTECTED]> wrote:
> You are correct. Apologies.
OK I guess. It would have taken way less time to re-check the site in
case you'd missed something than it took to flame me for trying to
help you...
Anyhow, the message has gone now, so I gues
Yep. I see that now.
When I went to the site, I just logged right in and ignored the line at
the top since I was in a hurry and just assumed it was an error on the
page. It is not formatted to the site or located where I would think a
outage notice would go so the eyes just blanked it out and k
On Wed, 2005-03-09 at 15:18 -0600, Matthew Boehm wrote:
> Again, may be off topic but are there any cards out there supported by
> asterisk that have on-board DSPs to do better 729->711 or 729->PRI
> conversion?
Not yet, and I don't know if anyone is working on the drivers for such a
card.
--
St
You are correct. Apologies.
Wiley
-Original Message-
From: Peter Bowyer [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 09, 2005 2:47 PM
To: Wiley Siler
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoIPJet
On Wed, 9 Mar 2005 14:40:17 -070
On Wed, 9 Mar 2005 14:40:17 -0700, Wiley Siler <[EMAIL PROTECTED]> wrote:
> No kidding. Really?
> Thanks for the help.
>
> As it were, I checked there already and there is no network status link.
> I sent an email to VoIPJet which has since been answered. However the
> problem was still occuring
On Wed, 2005-03-09 at 18:15 -0300, Jose Jorge Masdeu wrote:
> Hi, can I use an external modem such as Usr or other une attached at ttyS0
> port on my linux box whith asterisk for routing incoming and outgoing calls?
Simple search would have yielded many examples of why you can't.
External modems m
At the top of their homepage VoipJet has a one line posting:
Please note we are having a temporary glitch with our New York location.
Please send traffic to our West Coast Premium Server until the problem is
fixed sometime today. New SERVER IP: 69.25.60.30
I am sure that VoipJet like many others
Anyone tried the Clipcomm CG-410? I'm tired of beeing unable to get rid
correctly of the echo problem. I have 3 TDM04B installed in one server.
It was working fine when I had only one card installed in the server but
since I installed 2 more then I can't get rid of it what ever I try...
So now
No kidding. Really?
Thanks for the help.
As it were, I checked there already and there is no network status link.
I sent an email to VoIPJet which has since been answered. However the
problem was still occuring and I needed to see if it was just me or the
community at large.
Since nothing had
On Wed, 09 Mar 2005 13:06:24 -0800
Sean Kennedy <[EMAIL PROTECTED]> wrote:
Hi all, I'm running Asterisk 1.0.0. I am a customer (
and supporter ) of voicepulse. For me, it works
perfectly, but one of my customers noticed a small
problem: During a conversation, when the otherside isn't
talkin
Hi, can I use an external modem such as Usr or other une attached at ttyS0
port on my linux box whith asterisk for routing incoming and outgoing calls?
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Again, may be off topic but are there any cards out there supported by
asterisk that have on-board DSPs to do better 729->711 or 729->PRI
conversion?
-Matthew
TC wrote:
>> This maybe the wrong place to ask this question but... why did you
>> switch to the Sangoma?
> preliminary testing show Sango
On Wed, 9 Mar 2005 14:13:21 -0700, Wiley Siler <[EMAIL PROTECTED]> wrote:
>
>
> Anyone suffering an outage with them right now?
>
> I am getting the following from my box when I try to dial using themâ.
> == No one is available to answer at this time
Look at www.voipjet.com
--
Peter Bowye
Title: VoIPJet
Anyone suffering an outage with them right now?
I am getting the following from my box when I try to dial using them….
== No one is available to answer at this time
W
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Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter )
of voicepulse. For me, it works perfectly, but one of my customers
noticed a small problem: During a conversation, when the otherside
isn't talking, it's almost like the mic turns off.
Not that big of a deal I know, and t
Hi,
I see applications for signalling busy, congested, ringing, progress
etc, which I understand can be provided either in or out of band. But
all I want to do is generate a dialtone. This obviously can only be
done in band.
There is code for generating the tones when you have a physical line,
li
Hello!
I have a cabinet full of Wilcom Enhanced Line Powered Amplifiers with
Manual Balance, model ELPA-421V. I *believe* these were used for a bank
of analog modems back in the mid-90's. They were removed from a suite
when the old company moved out. Here's a URL:
http://www.wilcominc.com/e
thank you everyone!
It does not seen that it was configuration problems at all.
It appears it was the CVS that I was using from yesterday.
I decided to start over, downloaded the latest CVS, recompiled, and
voila! * started working
Indeed even a Cisco ATA that was never working before starte
On Wed, 2005-03-09 at 15:43 -0500, [EMAIL PROTECTED] wrote:
> For some reason I didn't think PostgreSQL was for mission critical apps. I
> don't think I have any reasoning behind it, just didn't think it was
> "hardcore"...sounds like i might be wrong...i'll have to look into it more.
>
> Open
Andrew Kohlsmith wrote:
On March 9, 2005 10:43 am, Cirelle Internet Products wrote:
How are you determining a fallback condition from one voip to another?
Mine's rather simple but it works well:
[macro-nufone-dial]
exten => s,1,GotoIf($[$ACCOUNTCODE != ""],s,gotac)
exten => s,n,SetVar(ACCOU
On Wed, 9 Mar 2005 15:43:46 -0500, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> For some reason I didn't think PostgreSQL was for mission critical apps. I
> don't think I have any reasoning behind it, just didn't think it was
> "hardcore"...sounds like i might be wrong...i'll have to look into i
ï
I also
have users that suffer from random echo on a British Telecom provided
PRI.
Can
you confirm that this has improved your user experience ?
Stuart
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Dennis
WebbSent: 09 March 2005 16:18T
Hi to all,
I'm using a TDM22B. When i establish an external call to the PSTN through an
FXO port, I'm not able to know the status of the call (no answer, busy, ...).
If I enable call progress (callprogress=yes) in Zapata.conf, I am able to
detect the no answer state but if the callee on the PST
For some reason I didn't think PostgreSQL was for mission critical apps. I
don't think I have any reasoning behind it, just didn't think it was
"hardcore"...sounds like i might be wrong...i'll have to look into it more.
Open source advantages are obvious, but aside from licensing and cost
fact
Hello,
We've released another update to our Asterisk GUI Client suite: 1.1.0
http://astguiclient.sf.net/
Screen shots: http://astguiclient.sourceforge.net/screenshots.html
The client suite runs on both Windows and UNIX, includes the VICIDIAL
auto-dialer and is free as in GPL.
(the suite is not
On Wed, Mar 09, 2005 at 02:13:02PM -0600, Dennis Webb wrote:
> Question about the make linux26 command. I use a 2.6 kernel and always
> do a straight make. Does adding the linux26 do anything except help the
> makefile know that it's a 26 kernel if it has trouble detecting for some
> reason? I m
> This maybe the wrong place to ask this question but... why did you
> switch to the Sangoma?
preliminary testing show Sangoma card/driver are better unload a full load
not such an issue with a single 4 span cards but 2+ cards
and the Digium T4xx cards start to drop calls, missed interupts etc
It might be more interesting to see how well it works with Xen--it's
supposed to be quite a bit faster then UML, *and* you can delegate
specific PCI devices to individual virtual systems. That means that
you could probably use Zap cards, not just SIP. I have no idea how
well it'd actually wor
any one know if asterisk and my IPH-90 work , configuring mgcp.conf like
this.
[general]
port = 2427
bindaddr = 172.20.10.5 ;asterisk addres
[172.20.10.15] ;IPH-90 ip address
context=local
host=172.20.10.15
line => aaln/1
thanks
___
Asteris
On Wed, Mar 09, 2005 at 02:50:47PM -0500, [EMAIL PROTECTED] wrote:
> I know this is a bit off topic but we are using Asterisk :)
>
> Since this list is full of tech gurus w/ all different sorts of
> backgrounds, I thought I would get the best opinions here.
>
> We have several different switche
Question about the make linux26 command. I use a 2.6 kernel and always do a straight make. Does adding the linux26 do anything except help the makefile know that it's a 26 kernel if it has trouble detecting for some reason? I might go ahead and try 1.0.7 and need to know if make linux26 does
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