You need to mention the DB deatils in sip.conf file .. please check
the WiKi pages
[general]
dbname=
dbhost=
dbuser=
dbpass=
Also you need to mention the DB details in res_mysql.conf
~Vamsi
On Mon, 14 Feb 2005 06:13:10 +, Jan Aabyevester
<[EMAIL PROTECTED]> wrote:
>
Does anyone know how to get a call back?
For example - you go to do an attended transfer, but nobody picks the call up.
You want to get the call back to say "Sorry, they are away from their desk"
Any idea what the key/combination/etc is?
Regards,
PaulH
-Original Message-
From: [EMAIL
Hi All,
I have a weird question regarding my Aastra 390 Analogue Phone.
Yesterday, I upgraded to the latest cvs (from feb 10th to the 13th).
This is when problems began. Now, I realize this probably has absolutely
nothing to do with CVS, but thought I should mention anyhow. I also
tried downgradin
Dear all,
My question is probably very trivial but Ill try my luck anyway.
Currently I have my asterisk PBX up and running but now I would like to perform real time SIP configuration. I have been reading the article Asterisk RealTime SIP, and been creating a mysql database called sipusers,
I use Fedora core 2, and openssl-0.9.7, expat-1.95.7 is installed by rpm packages.
I downloaded pwlib-1.5.2 and openh323-1.12.2 at my home directory(/root/root_src), asterisk 1.0.4 at directory /usr/src/ and have installed successfully.
Asterisk is executed normally, but module chan_h323.so c
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
> Sent: Friday, February 11, 2005 5:35 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Debian way of compiling zaptel kernel
> modules
>
> On Thu
You have what is a traditional 1A2 type telephone set. If you wish to
use the hold key and and lights you will need to connect a 1A2 KSU
between it and *.
Each line button has the following
pr 1 = T/R
pr 2 = A lead - this is a closure when the line is offhook
pr 3 = Lamp
Repeat x 4
Actually the
Hi,
I really appreciate the look and design of newer Mitel Ip phone.
I search througt the list and found only fews notes about the use Mitel 5055
phone on *. Anyone use other model (especially 52xx series) on * ??
Compatible? Easy to use? hassle to configure?
Thansk for your suggestion!
Bes
On Mon, 2005-02-14 at 13:52, Craig wrote:
> Message: 1
> Date: Mon, 14 Feb 2005 09:53:36 +1100
> From: "PHP Mechanic" <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Who makes these phones?
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Message-ID: <[EMAIL PROTECTED]>
>
Message: 1
Date: Mon, 14 Feb 2005 09:53:36 +1100
From: "PHP Mechanic" <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Who makes these phones?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; format=flowed; charset="iso-8
On Sun, Feb 13, 2005 at 08:35:33PM +0100, Michiel van Baak arranged a set of
bits into the following:
> On 20:08, Sun 13 Feb 05, Stefan Gofferje wrote:
> > Michiel van Baak schrieb:
> > > Thnx.
> > >No luck for me I guess.
> > >chan_sccp it will be.
> >
> > Not for the 79[05|12]... At least my 7
Folks,
I recently obtained a Western Electric multi-line phone and am
seeking help with getting this beast working with *.
The interesting stuff in my * implementation consists of a T100P
card, a TDM400P card, and an Adtran TA750 channel bank with three quad-port
FXS modules and a
On Feb 12, 2005, at 9:10 PM, Michael Giagnocavo wrote:
Hello,
I understand that only the destination of a call should do jitter
buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no
transfers), PhoneA and PhoneB need to perform their own jitter
buffering,
and Asterisk will just fo
Tzafrir Cohen wrote:
BTW: did I mention that we have binary packages for standard Debian
Sarge kernels in our apt source?
zaptel is the only package that never worked for me from apt-get. I need
to download, compile and install the kernel (specially because the
original debian install is pre 2.4.
Pure guess... you're probably bumping into some of the same issues
that many of us TDM users are hitting. Seems like either an interrupt
handling (latency) or pci bus issue. You'll find hundreds of postings
relative to this over the last six months or so. Not everyone has
problems with the TDM, but
- Original Message -
From: "Jon Gabrielson" <[EMAIL PROTECTED]>
You didn't say what your fxs/fxo requirements are but:
A T1 card ($500) and a used channel bank ($300) might be
a good alternative.
Basically my fxs/fxo requirements are the same as my existing TDM-400P ( 2
in 2 out). Just t
I had the same problems with Tormenta2 card from Digium.
Same behaviour, both cards were receiving irq`s, but when spans got up,
lots of messages ("Bad FCS") came up too on my asterisk console...
everything died with kernel panic in the end...
The motherboard was an Asus with dual Pentium3 933MHz
We have had a big success with the Linksys PAP2-NA. 2 FX ports and 1 WAN
port. Only downside is that only 1 call can be using 729 at a time. This has
been confirmed with Linksys. They will be releasing PAP2-NAv2 in March to
overcome this. In the meantime, get a Sipura 2100, supports 2 729 calls and
For whatever it's worth, it was the crossover cable.
--- Andrew Kohlsmith <[EMAIL PROTECTED]>
wrote:
> On February 12, 2005 09:21 pm, David Coulson wrote:
> > If he gets a green light with a loopback plug
> wired like that, his
> > controller is definatly screwed up :-)
> >
> > 1->4
> > 2->5
> >
> > For planning purposes, is it appropriate to think in terms of
> > purchasing
> > a t38 capable box even if its not supported by * today? (I'm well aware
> > of the bounty and Steve's work.)
>
> That's what I would do. In fact, I already have T.38 capable VOIP
> adapter (an Azatel 200) for my
> I spent a bunch of time troubleshooting the SIP end of things, thinking
> that's where the
problem was, until I realized that every other SIP
> connection I make (from remote) yields a high quality call. ie. I can dial
> another SIP
client and maintain high quality audio. Additionally, I
You didn't say what your fxs/fxo requirements are but:
A T1 card ($500) and a used channel bank ($300) might be
a good alternative.
You also might check out the voicetronix cards.
Cheers,
Jon.
On Sunday 13 February 2005 02:55 pm, Paul Fielding wrote:
> Are there any other relatively low cost
Hi,
Telecoms in China is not based on American standards. It is based on
European standards. IDN in China is exactly the same as ISDN in Europe,
and European configurations on Asterisk will work in China.
Regards,
Steve
Xu, Duo wrote:
Hi,
I plan to install asterisk and connect it to telco
thr
Should be a good night - looking forward to seeing some unfamiliar faces!
Regards,
PaulH
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jurgen
Sent: Thursday, 10 February 2005 12:55 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Melbo
On Mon, 2005-02-14 at 10:10, Gary wrote:
> On Mon, 14 Feb 2005 09:53:36 +1100, PHP Mechanic wrote:
>
> >http://www.broadbandphone.com.au/global/pnp.htm
>
>
> They look like they are all PA1688 based.
The black one is a dead copy of the one sitting on my desk, made by
Hirakawa Electronics accord
Additionally when I do receive the unreachable message as soon as I
place an outbound call the peer becomes reachable..
dabigshiznizzle wrote:
I have been playing with asterisk for a couple of weeks now and I
have been very happy with its performance. However, I have run into a
problem
On Mon, 14 Feb 2005 09:53:36 +1100, PHP Mechanic wrote:
>http://www.broadbandphone.com.au/global/pnp.htm
They look like they are all PA1688 based.
Gary
.
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http://www.broadbandphone.com.au/global/pnp.htm
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It is 0* + number not 0 + number only
other way is to use a gatekeeper and register the asterisk and the polycom to
it..
In my h323.conf
[4500]
type=user
host=10.10.10.59
context=default
in my extensions.conf
[h323]
exten => 4200,1,Dial,H323/10.10.10.49
exten => 4300,1,
At 03:36 PM 2/13/2005, you wrote:
On 21:47, Sun 13 Feb 05, Vikram Rangnekar wrote:
> +++ Michael Devenijn [13/02/05 18:23 +0100]:
> >
> > We got also these problems and where searching like fools for solutions
> > ... until the time we changed the main board of the server! (Interrupt
> > sharing or
I have been playing with asterisk for a couple of weeks now and I
have been very happy with its performance. However, I have run into a
problem with how I want to deploy this solution.
I have a mix of softphones (SJ and Xlite), ATA's, and a couple of IP
phones (Snom 190). The asterisk box
Rich Adamson wrote:
<> This seems to be a problem with the
current wctdm driver. It seems to be broken for audio going out. I used
to be able to send faxes reliably using spandsp and a TDM40P card, but
I no longer can. I haven't had time to look in detail at what is wrong.
I'd lo
On 21:47, Sun 13 Feb 05, Vikram Rangnekar wrote:
> +++ Michael Devenijn [13/02/05 18:23 +0100]:
> >
> > We got also these problems and where searching like fools for solutions
> > ... until the time we changed the main board of the server! (Interrupt
> > sharing or Hyper threading stuff, I don't r
On Sun, 2005-02-13 at 18:10 +0100, Andres GÃmez GarcÃa wrote:
> Thanks Bruno, I'll try it.
Also, you might take a look again at
http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=272259
Following your mail, I wrote to that list (cf the last mails there),
and it looks like a working oh323 package w
Hi,
my success story with the zaphfc incl. florz patch has been to early.
Allthough sound drop outs no longer happen, the following happens after
a longer period (2 days) of inactivity on the asterisk box.
Feb 13 22:30:15 NOTICE[6799]: chan_zap.c:7685 pri_dchannel: PRI got
event: HDLC Abort (6)
I'm currently deciding on what card to pruchase for octo/quad BRI card to
use with Asterisk on EuroISDN lines.
I'm aware of at least two options (Junghanns or Beronet), but don't know how
stable and well supported they are. Which ones are better supported ? Any
experiences? Any advice ? How tos ?
U
On Feb 12, 2005, at 4:38 PM, Rich Adamson wrote:
For planning purposes, is it appropriate to think in terms of
purchasing
a t38 capable box even if its not supported by * today? (I'm well aware
of the bounty and Steve's work.)
That's what I would do. In fact, I already have T.38 capable VOIP
adap
Are there any other relatively low cost analog
cards available? I'm interested in finding something that might work a bit
more reliably than the TDM-400P
regards,
Paul
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I'm running a TDM-400P with 2 x FXS and 2 x
FXO. I'm finding that there seems to be an odd relationship to
sound quality on the card to my local when connecting via a SIP
client.
When I'm on my local network, if I connect to
Asterisk via a SIP client (such as x-pro), and dial an outside
I have set Asterisk as a gateway on the Polycom and set gatekeeper to
"No"
So to dial on the Polycom I would then dial (0+the number). No way to
just dial directly without the 0?
The other side of this is how do I dial "to" the Polycom, I have tried
everything that I can think of for the "e
+++ Michael Devenijn [13/02/05 18:23 +0100]:
>
> We got also these problems and where searching like fools for solutions
> ... until the time we changed the main board of the server! (Interrupt
> sharing or Hyper threading stuff, I don't remember) we replaced the
> Supermicro board with an intel.
> This seems to be a problem with the current wctdm driver. It seems to
> be
> broken for audio going out. I used to be able to send faxes reliably
> using spandsp and a TDM40P card, but I no longer can. I haven't had
> time
> to look in detail at what is wrong
On 20:08, Sun 13 Feb 05, Stefan Gofferje wrote:
> Michiel van Baak schrieb:
> > Thnx.
> >No luck for me I guess.
> >chan_sccp it will be.
>
> Not for the 79[05|12]... At least my 7905 does not like chan_sccp too
> much and they crashed my * (1.0.5)... unless you bounty the chan_sccp
> developer
I’d be grateful if someone could point me in the right
direction.
I have a Broadvoice trunk attached to Asterisk which I use
for frequent calls to the UK
using the following in extensions.conf
exten =>
_0[1-68].,1,Ringing
exten =>
_0[1-68].,2,Dial(SIP/BV/01144${EXTEN:1})
exten =
Hey guys,
I got moh-native working with today’s CVS of asterisk
and asterisk-addons so I’m guessing there were some code problems with
versions 1.0.1, 1.0.4 and current CVS stable. Following the wiki
instructions worked fine. Also the mp3’s that come with Asterisk
sound perfect, where
Michiel van Baak wrote:
On 04:06, Mon 14 Feb 05, Shaun Ewing wrote:
On Sun, 13 Feb 2005 12:58:47 +0100, B. Vallet -
www.acropolistelecom.net <[EMAIL PROTECTED]> wrote:
Here it is :
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905
software is the same for 7905 / 7912
When I go to that url i
Anton Krall wrote:
How do you make SIP work behind NAT without having to change anything on the
firewall for example, those cable modems
So far, Ive tested this using softphones and only iaxphone has been able to
work using IAX, eye lite or something for FWD that uses SIP says it cant
connect t
Jim Van Meggelen wrote:
> Yeah, I can't see that working. 0.0.0.0 isn't really an address; more
like a lack of one. That's what's got me wondering about DHCP.
The IAXy does not use DHCP, it uses the older BOOTP protocol. Most
DHCP servers support BOOTP (but it may have to be enabled)
___
Hi list!
I'm not getting incoming CallerID in The Netherlands on my TDM11B.
Everything was configures according to the docs at digium.com.
The error on the console is this:
Feb 13 16:49:40 ERROR[16123]: callerid.c:260 callerid_feed: fsk_serie made
mylen < 0 (-84)
Feb 13 16:49:40 WARNING[16123]:
> -Original Message-
> From: Luki [mailto:[EMAIL PROTECTED]
>
> The Sipuras have a ton of configurable parameters. If you
> understand them (and there is no good manual, unfortunately)
Really? 87 pages aren't enough for you?
http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pd
Hi,
I plan to install asterisk and connect it to telco
through ISDN in China.
I'd love to know if the ISDN standard in China has any
difference than in America before I buy the digium
card.
anybody has experience in it? or anybody who installed
asterisk with ISDN in asia can share their
expier
Sascha E. Pollok wrote:
Good evening,
allow me to join in right here. Which ATA/TA would you
suggest for connecting analogue fax machines to Asterisk?
One of the ones named before or e.g. a ATA-186 made by Cisco?
At the moment I am deploying Grandstream ATAs for faxing machines with
out a problem
Thanks to everyone who responded. I submitted a bug
report to digium
(http://bugs.digium.com/bug_view_page.php?bug_id=0003580),
and markster responded, suggesting that I get an
updated version of stable asterisk from CVS. I did,
and now it's all working fine. I must have initially
downloaded a not-
Good evening,
allow me to join in right here. Which ATA/TA would you
suggest for connecting analogue fax machines to Asterisk?
One of the ones named before or e.g. a ATA-186 made by Cisco?
Cheers
Sascha
> The Sipuras have a ton of configurable parameters. If you understand
> them (and there is n
On 19:36, Sun 13 Feb 05, Stefan Gofferje wrote:
> Michiel van Baak schrieb:
> >On 04:06, Mon 14 Feb 05, Shaun Ewing wrote:
> >
> >>On Sun, 13 Feb 2005 12:58:47 +0100, B. Vallet -
> >>www.acropolistelecom.net <[EMAIL PROTECTED]> wrote:
> >>
> >>>
> >>>Here it is :
> >>>
> >>>http://www.cisco.com/cgi
Heh. Good point, Kevin. I didn't realise that ast_data
was also a third party add-on. :)
So I submitted a bug report to digium with my gdb
trace
(http://bugs.digium.com/bug_view_page.php?bug_id=0003580),
and markster there suggested that I should update to
the latest stable asterisk from CVS. I di
The Sipuras have a ton of configurable parameters. If you understand
them (and there is no good manual, unfortunately) then you can be of
great benefit. Otherwise they'll be worthless. I particularly miss the
dial-plan, distinctive ring and audio gain options on the
Grandstreams. Remote syslog can
Hi,
I am thinking of purchasing a cheap Dlink VPN for
testing purposes for use with my Asterisk box and would like to ask the list for
advice on how to pick a VPN that will work with my box. I am a newbie to both
VPN's and Asterisk so any advice will be appreciated.
Thanks,
Mike
___
Anton Krall wrote:
Guys.. which ATA is better for connecting analog phones (features,
stability, experiences, etc)?
Sipura 2000 or Handy Tone 286, etc?
What are you experiences?
In my experience the Sipura 2000 has three hardware advantages:
* 2 independent phone ports
* Mounting holes
* The p
I found out the CD's I made were OK - I used one on a different computer
and it worked fine.
[EMAIL PROTECTED] doesn't like the current Asterisk box I'm using now. It's
an IBM Netfinity 3500 - dual 233MHz processor, SCSI, 512MB, DVD-ROM,
blah,
blah. That's the only computer I get the error me
On Sat, 12 Feb 2005, Michael Giagnocavo wrote:
> Hello,
>
> I understand that only the destination of a call should do jitter
> buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no
> transfers), PhoneA and PhoneB need to perform their own jitter buffering,
> and Asterisk
Forrest W. Christian wrote:
Actually, that isn't quite 100% accurate.
And even yours wasn't 100% accurate. Instead of messy extension lines
you could setup a Queue as well.
Flexibility, this is why Asterisk rules!
Jeremy McNamara
___
Asterisk-Users
On Fri, 11 Feb 2005, Brian Buhrow wrote:
> Hello. You can't have two phones login with the same extension. You
> need to assign one phone to 101, and the other to 102. Set the user to 101
> on one and 102 on the other.
Actually, that isn't quite 100% accurate.
The more accurate statemen
> So, which way to go? IAX or SIP? IAXy or Sipura?
I prefer by far IAX
> All ip phones use SIP right?
Nope, now there's IAX hardphone, like there : http://www.iaxtalk.com/
hth
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Rich Adamson wrote:
Steve,
Need to replace our older soho fax machine with something more current.
Would like to run the fax line through *, but haven't been able to
make spandsp work correctly with digium TDM04b card. Our fax volume
This seems to be a pr
Guys.. which ATA is better for connecting analog phones (features,
stability, experiences, etc)?
Sipura 2000 or Handy Tone 286, etc?
What are you experiences?
__
Anton Krall
___
Ast
Yes, but I have to configure a route for each host in every host! A the
moment i have about 120 Asterisk hosts and every astersk have about
50-100 users! Is for that I want a single sip proxy that route dial.
I read more about ser, and the suggestion is to use ser for accounting
and route, and aste
On 04:06, Mon 14 Feb 05, Shaun Ewing wrote:
> On Sun, 13 Feb 2005 12:58:47 +0100, B. Vallet -
> www.acropolistelecom.net <[EMAIL PROTECTED]> wrote:
> >
> >
> > Here it is :
> >
> > http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905
> >
> >
> >
> > software is the same for 7905 / 7912
W
We got also these problems and where searching like fools for solutions
... until the time we changed the main board of the server! (Interrupt
sharing or Hyper threading stuff, I don't remember) we replaced the
Supermicro board with an intel.
Try the same config on another machine (maybe an older
How do you make SIP work behind NAT without having to change anything on the
firewall for example, those cable modems
So far, Ive tested this using softphones and only iaxphone has been able to
work using IAX, eye lite or something for FWD that uses SIP says it cant
connect to the provider...
On 12 Feb 2005, at 19:46, eric m wrote:
Hi!
I would like to have feedback on wireless (wifi / 802.11b) IP phone to
use
with Asterisk PBX. Can you sugest model, The best and also the worst
to
use.
I've been using the Zyxel P2000 for a month or so now.
I was going to deploy several of them around
El dom, 13-02-2005 a las 16:57 +0100, Bruno Hertz escribió:
> Had the same issue with Debian Sarge. I didn't actually investigate it,
> but I strongly suspect the openh323/pwlib packages don't work with the
> asterisk-h323 package. The H323 README specifically says btw to don't
> use the packages o
On Sun, 13 Feb 2005 12:58:47 +0100, B. Vallet -
www.acropolistelecom.net <[EMAIL PROTECTED]> wrote:
>
>
> Here it is :
>
> http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905
>
>
>
> software is the same for 7905 / 7912
It's not actually.
Firmware for both versions is available from t
[EMAIL PROTECTED] wrote:
>> asterisk-users-bounces at lists.digium.com wrote:
>>> Hi everyone,
>>>
>>> I was working yesterday and after I provide my IAXy box it loose any
>>> network comunication, the link light (green) is on and the activity
>>> light (orange) when the power is turned on it does
Hi Guys,
I’ve attempted to get this moh-native thing to work
with no success. I’ve reviewed wiki, mantis and e-mail postings and
I’m confused.
The latest I’ve read is native moh should be in
asterisk-addons in format_mp3, but what version will it work with? I’ve
tried asterisk 1.0
On 11:52, Sun 13 Feb 05, Mike Clark wrote:
> Michiel van Baak wrote:
>
> >On 10:56, Sun 13 Feb 05, John Middleton wrote:
> >
> >
> >>Has anyone any experience of the above.
> >>Key feature for me is tracking incoming and outgoing emails and
> >>linking them to the contact record.
> >>
> >>Thanks,
beonice wrote:
I don't know whether RealTime PostgreSQL, but I can't
upgrade to RealTime anyway ... I need a stable version
of asterisk, and the current stable version does not
include RealTime. :(
You need a "stable" version of Asterisk, but you're willing to patch
with an unsupported change like
Michiel van Baak wrote:
On 10:56, Sun 13 Feb 05, John Middleton wrote:
Has anyone any experience of the above.
Key feature for me is tracking incoming and outgoing emails and
linking them to the contact record.
Thanks, sorry for the OT ;-)
Hi,
Have a look at http://www2.covide.net
Maybe tha
Addendum: I did a little investigation and found this
http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=272259
Regards, Bruno.
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To UN
Roger Hanson wrote:
I've downloaded 2x and burned 2 cds and get an error "invalid
compressed format (err=2) system halted" message both times.
It'd be nice to have a MD5 to verify my download is OK. It'd narrow
down the problem to either the download or the burn, wouldn't it?
Here is an _un-off
On Sun, 2005-02-13 at 04:46 +0100, Andres GÃmez GarcÃa wrote:
> I've tried GNOMEMeeting also. It works fine with a P2P client
> connections (ALSA works fine) but, even when I success connecting to an
> asterisk server, I haven't hear anything. I mean, I don't hear the demo
> successfull messages.
Oh yeh, their database admins have been playing funny games with the
rules. It's been demonstrated on more than a few 'key words'
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Duane
Sent: Sunday, February 13, 2005 10:25 AM
To: Asterisk Users Mailing Lis
> Try using "context" (with a trailing T!!) in your config, and lose the
> spaces around the equal sign, just in case.
Well, I was wondering why the error log showed that the phones where
in default context.
That just show that I should never answer before my first coffee ;-)
On Mon, February 14, 2005 2:18, dean collins said:
> The limited domain reference is obsolete, Telstra have a 2 million
> record database (yeh I know it's a lot smaller when you dice it
> phonetically but it's still big enough).
Maybe it's just me, but I found their database very hit and miss, no
--- Andrew Kohlsmith <[EMAIL PROTECTED]>
wrote:
> On February 12, 2005 07:31 pm, Richard Reina wrote:
> > On thing that is odd is that although the t1 cross
> > over cable is plugged in to both * and the Adit.
> Both
> > t1 and t1 leds on the Adit are red. How can they
> both
> > have the same
The limited domain reference is obsolete, Telstra have a 2 million
record database (yeh I know it's a lot smaller when you dice it
phonetically but it's still big enough).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Underwood
Sent: Sunday, Febr
Try using "context" (with a trailing T!!) in your config, and lose the
spaces around the equal sign, just in case.
> -Original Message-
> From: Wesley Jay Deypalan [mailto:[EMAIL PROTECTED]
> Sent: Saturday, February 12, 2005 9:33 PM
> To: [EMAIL PROTECTED]; asterisk-users@lists.digium.co
Hi Dean,
You seem to have had your head up the supplier's arse for a number of
years. :-)
I last tried a Nuance demo system in about 2002, and found it useless.
Speechworks (now scansoft) was rather better, but still useless for
English. I'm British. Trying the British system gave poor results.
Thank you for the response ... Nicolas (the author of Flash Panel) had
responded with this too, but you have to be using 0.20-unstable, where as I
was using 0.19-stable.
I have 0.20-unstable running and the park button works for the most part -
seems to stay "lit" even after parking times out, but
Matthew,
I believe you're the original developer on the
RealTime code ... do you know if it will work with
Asterisk 1.0.5 or whatever the stable version of
Asterisk is?
Thanks,
Maya
--- Matthew Boehm <[EMAIL PROTECTED]> wrote:
> Why not just use the built-in database features to
> do what you
Lonnie,
If you look at:
http://www.voip-info.org/wiki-Asterisk+RealTime
it says that MySQL _is_ supported.
I don't know whether RealTime PostgreSQL, but I can't
upgrade to RealTime anyway ... I need a stable version
of asterisk, and the current stable version does not
include RealTime. :(
I a
John Middleton wrote:
Has anyone any experience of the above.
Key feature for me is tracking incoming and outgoing emails and
linking them to the contact record.
Thanks, sorry for the OT ;-)
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Steve,
> >Need to replace our older soho fax machine with something more current.
> >Would like to run the fax line through *, but haven't been able to
> >make spandsp work correctly with digium TDM04b card. Our fax volume
> >
> This seems to be a problem with the current wctdm driver. It seems to
Steve then you have had your head up your arse for a number of years.
Nuance was delivering 90% in 1999 and I have a number of happy customers
to prove it.
You also obviously didn't look at either the Nuance or angel sites
because both of them offer free form speech to text capabilities.
One of
On 21:21, Sat 12 Feb 05, Robert Rozman wrote:
> Hi,
>
> could you give some more info about your setup. How do you get 2 fritz cards
> working (I thought it works only on 2.4 kernels ) ?
>
> What capi drivers do you use ?
>
> Thanks,
>
> regards,
>
> Rob.
>
Hi,
I followed the instructions in
On 10:56, Sun 13 Feb 05, John Middleton wrote:
> Has anyone any experience of the above.
> Key feature for me is tracking incoming and outgoing emails and
> linking them to the contact record.
>
> Thanks, sorry for the OT ;-)
Hi,
Have a look at http://www2.covide.net
Maybe that's what you want.
On 14:46, Sat 12 Feb 05, eric m wrote:
> Hi!
>
> I would like to have feedback on wireless (wifi / 802.11b) IP phone to use
> with Asterisk PBX. Can you sugest model, The best and also the worst to
> use.
>
> Thanks,
>
> eric.
Hi,
I read on sf that the cisco wireless phone is almost 100%
worki
Sjaak Nabuurs wrote:
> Hello * users
>
> I've problems with sound quality on zaphfc
> Asterisk works fine good sound quality.
> If I do "make load" in the bristuf.xx zaphfc dir then sound quality
> drops directly.
> Even if I don't load the chan_zap in the modules.conf
>
> I use this config on
Duane wrote:
On Sun, February 13, 2005 23:19, Vikram Rangnekar said:
I'm sorry i didnt quite understand what you meant why would i need 4
d-channels i've only used 16 and 47 as my dchannels and want span 1 to
generate the clock for this e1 setup.
As far as I'm aware each E1 has 30 b channel
On Sun, February 13, 2005 23:19, Vikram Rangnekar said:
> I'm sorry i didnt quite understand what you meant why would i need 4
> d-channels i've only used 16 and 47 as my dchannels and want span 1 to
> generate the clock for this e1 setup.
As far as I'm aware each E1 has 30 b channels, and 2 d c
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