Hello All,
This is my first post to this list, I hope i'll get my problem solved
here.
I was using following set of software for my network within the LAN.
linphone[as SIP UA],partysip[SIP PROXY SERVER],GNUosip[SIP STACK]. with
ALSA SOUND DRIVERS.
Then I wanted to make IP-PSTN and PSTN-IP calls,
I need some tips on configuration of voicemail with mysql...
here is my voicemail.conf
**voicemail.conf***
[general]
dbhost=localhost
dbname=asteriskvmusers
dbuser=root
format=wav
serveremail=asterisk
attach=yes
maxmessage=60
maxgreet=60
I cannot get the Message Waiting Light (MWL) on my Grandstream phone to turn
on when I leave a new voice mail message for that phone. I have specified
the correct mailbox in my sip.conf as follows:
[200]
type=friend
username=200
host=dynamic
context=dialout
callerid=200
dtmfmode=rfc2833
mailbox=20
On Thu, 26 Feb 2004, Rana Dutt wrote:
> Also, I find it disconcerting that there's a Conference button on the
> Grandstream phone, but when it's pressed, nothing happens. If this sends out
> some sort of switch-hook flash, can Asterisk intercept it, and then use the
> meetme app? Do the Cisco phon
Thanks for the info. Which phones support consultation transfers? The
Grandstream and IpDialog phones most certainly do not.
Also, I find it disconcerting that there's a Conference button on the
Grandstream phone, but when it's pressed, nothing happens. If this sends out
some sort of switch-hook f
Alas I feel the same way, but my service provider has the 800 numbers
connect to the POP via a Lucent Switch and then pass the 800 calls to a
local inter connect via some other switch. Ask your provider if they
can pass you the actual DNIS. In my case Verizon (Philadelphia) can
but my lower rate
are you on a machine that is slow or running alot of stuff? The ongoing
answer is the thread that is run by asterisk can't complete it's task
fast enough due to lack of system resources so it creates the notice
below.
On Wed, 2004-02-25 at 20:55, Carl Lougher wrote:
> When I call Voicemail I get a
Hi List,
how does one route calls in extensions.conf via DNIS ??
I need to route the 800 number that was dialed to the
right part inside of asterisk. I don't want to waste
a PSTN DID for each Watts number.
thanks
___
Asterisk-Users mailing list
[E
rumor has it that new firmware will let you adjust the
ringer voltage on HT-286 / HT-486
On Mon, Feb 23, 2004 at 09:41:24AM +0100, Nicolas Bougues wrote:
> Dear all,
>
> My GS ATA-286, which otherwise work well, seem to be unable to ring a
> fax (or at least, some kind of fax). The fax basical
I've already got an ISDN line that I use for my IP connection at home.
I'd like to try out * with ISDN, so I'm shopping for a card. I'm in the
US, so I need a U interface, not S/T. Can anybody make any suggestions?
An awful lot of the cards supported by isdn4linux are hard or impossible
to find w
When I call Voicemail I get a very slow underwater sounding voice for the
first few seconds then it corrects itself. Any idea?
Output from Console:
-- Executing VoiceMailMain("SIP/2101-20db", "") in new stack
-- Playing 'vm-login' (language 'en')
Feb 26 14:45:58 NOTICE[393234]: sched.c:218 s
Ok I changed some stuff around and got it to work.
Now the strange part... The caller if they press # will ask ME who id
like to transfer to? :)
So if they accidently hit #, they get put on hold
Any ideas? :)
Sincerely,
Matthew Marlowe
Gear 3 Technologies, LLC
609.252.1155 x614
www.gear3.com
Having a problem with call forwarding If I call into the main number
go through the auto attendant and choose the persons extension it
forwards out to there alt number they specified. But if call them
directly via there DID... The call rings back the person calling the
DID.. Dosen't make since
On Tue, Feb 24, 2004 at 10:37:21PM -0500, Tim Sailer wrote:
> On Tue, Feb 24, 2004 at 04:33:41PM -0500, mattf wrote:
> > Take a look at my GUI app:
> >
> > http://sourceforge.net/projects/astguiclient/
> >
> > It'll run on Linux and Windows, it's written in perl and it'll list every
> > channel(Z
On Wed, Feb 25, 2004 at 05:46:55PM -0500, Andrew Thompson wrote:
> > I looked at it, and it won't work for me the way you have written it
> > without me hacking at it. You insist on having MySQL run in network
> > mode. I really don't want another port/service running on a box that
> > is internet-
Hi,
In the Makefile inside asterisk/channels/h323 directory, there's a line like
this:
CFLAGS += -I$(PWLIBDIR)/include/ptlib/unix -I$(PWLIBDIR)/include
try to use "-I$(PWLIBDIR)/include" ONLY, it should work. I've compiled it
with pwlib 1_6_2, which works fine
leo
-Original Message-
F
I'm running * with a 7960/Skinny. I'm seeing several pages with
SIPDefault.cnf config file, but as I'm not running SIP for this phone
(yet), it is useless now.. Is there any Skinny/SCCP Default.cnf also?
Actually, I'm trying to enable the extra 7960 features, like directory,
services etc...
Thanks
I conference calls all the time. Asterisk is telling you what the problem
is. You are running a codec that it doesn't like for whatever reason.
I use ULAW and ALAW with absolutely no problem.
Chris Clifton wrote:
Is app_meetme the only way to conference calls on a 7960 with * ? It looks
as if
yes I know two Web based iax client.
the one is
Babar Shafiq's iaxclient.ocx
http://www.geocities.com/babarnazmi/index2.htm
and another one is
Dan's activediax
http://www.laser.com/dante/diax/diax.html
so usefull :-)
We can find topics at here.
http://iaxclient.sourceforge.net/
mack_jpn
O
>
> I tried at first Theo's chan_sccp. (No Dialtone)... I have now tried the
> lambda-solution and asterisk keeps crashing every time I have used the
> 7912... I guess that skinny / asterisk have some way to go before it is of
> any use other than playing with.
please do start asterisk with safe_
On Wed, 25 Feb 2004, mattf wrote:
> It actually has components that run on the Asterisk server as well as a GUI
> app that runs on the desktop(Win32 or Linux).
>
> astguiclient was really designed for a larger environment than most of the
> Asterisk client apps that are out there, and it is not a
On Wed, 25 Feb 2004, Matthew B Marlowe wrote:
> Greg,
>
> When using your patch, when the file is being played... The MOH stops
> for the caller while that sound file is played to me. Then the music on
> hold continues and if I hit # the music on hold stops once again and
> let's me talk.
>
> I
On Wed, 25 Feb 2004, Matthew B Marlowe wrote:
> Post your configs, maybe we can help. :)
I did. In my original post. Along with the original question. I don't
think people are reading properly.
My original post is archived here:
http://lists.digium.com/pipermail/asterisk-users/2004-February/0
On Wed, Feb 25, 2004 at 06:19:13PM -0500, Matthew B Marlowe wrote:
> You said using the # feature gives you the ability to not lose a call if
> you dial an invalid extension but that doesn't work for me.
>
> When I dial # and enter an invalid extension I want it to sy ' invalid
> extension' to ME
You said using the # feature gives you the ability to not lose a call if
you dial an invalid extension but that doesn't work for me.
When I dial # and enter an invalid extension I want it to sy ' invalid
extension' to ME and ask me to reenter one.
Instead if I dial # (or transfer button) and ente
It actually has components that run on the Asterisk server as well as a GUI
app that runs on the desktop(Win32 or Linux).
astguiclient was really designed for a larger environment than most of the
Asterisk client apps that are out there, and it is not as easy to set up. It
was also initially desig
> On Thu, Feb 26, 2004 at 09:43:00AM +1100, Adam Hart wrote:
> > Wierd errors, the actual library compiled fine though? Cause pdirect.h
> > doesn't been touched for 5 months
>
> Yup, PWlib and OpenH323 built fine, no errors.
If you're game, try and work out why openh323's Simple compiles but ast_h
On Wednesday 25 February 2004 05:35 pm, Matthew B Marlowe wrote:
> How'd you get the # transfer feature working? :)
My transfer button worked out of the box. Then when I upgraded * in
Nov-Dec it stopped working I have to use #.
>
>
> Sincerely,
> Matthew Marlowe
> Gear 3 Technologies, LLC
> 609.
On Thu, Feb 26, 2004 at 09:43:00AM +1100, Adam Hart wrote:
> Wierd errors, the actual library compiled fine though? Cause pdirect.h
> doesn't been touched for 5 months
Yup, PWlib and OpenH323 built fine, no errors.
___
Asterisk-Users mailing list
[EMAIL
Looking at the reference design for the chipset used in an X100P a fair
chunk of capacitance is slapped straight across the line which would
present a significant load to DMT signals. I guess the fax machine
introduces some inductance in series with the phone to compensate.
I found this link t
[EMAIL PROTECTED] wrote:
> We are using IAXphone in a production environment with hardphone
> backup (BRI).
>
> Most things work nicely, but we have found that we can only transfer
> each call once.
>
> I.E. Incoming call comes in on the [incoming] context, receptionist
> transfers it to a loca
Wierd errors, the actual library compiled fine though? Cause pdirect.h
doesn't been touched for 5 months
- Original Message -
From: "Jim Rosenberg" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, February 26, 2004 9:33 AM
Subject: Re: [Asterisk-Users] Patching Asterisk for Ope
> I looked at it, and it won't work for me the way you have written it
> without me hacking at it. You insist on having MySQL run in network
> mode. I really don't want another port/service running on a box that
> is internet-facing... call me silly. :) For things like this, I'll
> only have the se
Ok I'm sorry I got # to work to transfer calls but if you enter an
invalid extension, how do you have asterisk handle that?
Sincerely,
Matthew Marlowe
Gear 3 Technologies, LLC
609.252.1155 x614
www.gear3.com
(00)
>< Choose a job you love, and you will
/||\ never have to work a day in
How'd you get the # transfer feature working? :)
Sincerely,
Matthew Marlowe
Gear 3 Technologies, LLC
609.252.1155 x614
www.gear3.com
(00)
>< Choose a job you love, and you will
/||\ never have to work a day in your life.
=/\=
-Original Message-
From: [EMAIL PROTECTED]
[mailto
On Thu, Feb 26, 2004 at 09:13:20AM +1100, Adam Hart wrote:
> Yes, asterisk is vulnerable if you have H.323 running.
> What happens when you try and compile asterisk with the latest version of
> OpenH323, it's been a few months since i've done it but it used to work.
A flood of errors. Starting of
On Wed, 25 Feb 2004 11:58:54 -0700, [EMAIL PROTECTED] wrote:
>You may be right here. I was thinking of an ActiveX plug-in. I don't expect them to
>use public internet kiosks so they should be able to use the ActiveX approach. I was
>hoping that something IAX based could be found as it would mak
> The consensus in the Asterisk community seems to be that (somehow)
Asterisk
> is not vulnerable to these security holes, which many experts consider
> quite serious. I am frankly having a lot of trouble understanding where
> this bliss is coming from. From my reading on this, it looks to me as
>
The Grandstream BudgeTone 101 phone has a Transfer button. This appears to
be a "blind" transfer: once you've dialed the extension to which you want
to transfer, the phone tries to do this and then "dumps you out".
My question is this: Let's say I explain to my users that I don't want
them using t
Robert Sprockeels a écrit :
Jean-Denis,
Your suggestion works for me indeed. I had to test around a little, and
I had to put the entire MSN number in the extension field. How come you
only put part of it in there? Did you specify something about the common
part in the modem.conf file?
Nothing spec
I need to know how to get Asterisk patched for the recent vulnerabilities
in various H.323 implementations due to integer overlows in ASN.1 parsing.
I'm quite new to this world of Asterisk, H.323, SIP, and VoIP, so please
bear with me if I garble something.
The consensus in the Asterisk community
I have commented de Republica2 in sip.conf 'cause if I uncomment it neither
Republica and Republica2 register (maybe because they're on the same
gateway?)
Well, inspite it register well when I try tocall any extension It plays
'busy' tone immediately after Asterisk takes the calls I thought it was
--On Friday, January 23, 2004 12:03 PM +0100 Philipp von Klitzing
<[EMAIL PROTECTED]> wrote:
There are also - less established - ways to manage sip.conf,
extensions.conf and voicemail.conf with the help of mySQL (or some other
database backend).
Any more details on this?
__
Seems like republica registers ok, but not republica2. Republica2 failes to authenticate.
You have a normal registration sequense here:
-Client sends a REGISTER without authentication
-Server sends "trying..."
-Server sends 407 Proxy auth (should be WWW auth) with challenge
-Clients ACK
-Client s
At 12:34 PM 2/25/2004, you wrote:
On Wed, 25 Feb 2004, Ernest W. Lessenger wrote:
> With recent CVS builds I've been able to specify 7000 and 7001-7200 as the
> call parking lot. I haven't tried any other numbers.
The "parking lot" is assigned by the user or by the system?
I found that my * is ass
On Wed, 25 Feb 2004, Ernest W. Lessenger wrote:
> With recent CVS builds I've been able to specify 7000 and 7001-7200 as the
> call parking lot. I haven't tried any other numbers.
The "parking lot" is assigned by the user or by the system?
I found that my * is assigning 'lot' 701 for my "parked c
On this cuts note that the gateway has username 'Republica', you could see
some reference to Republica2 which corresponds to a second line on the
gateway that I have disabled.
Thanks for your help!
That's SIP debug when dialling '9' (9 would do Goto(s,1))
===
*CLI>
*CLI>
11 hea
At 11:48 AM 2/25/2004, you wrote:
No matter what I put in parking.conf for parkpos, I find that the first
call is always parked on 701. Is this a bug?
With recent CVS builds I've been able to specify 7000 and 7001-7200 as the
call parking lot. I haven't tried any other numbers.
--Ernest
Did anybody make X-web Lite to work with Asterisk ?
It seem I cannot make it work.
- Original Message -
From: "Philipp von Klitzing" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, February 25, 2004 4:39 PM
Subject: Re: [Asterisk-Users] Web based UA
> Hi!
>
> > Does anyone
Title: Calls always parked on 701
No matter what I put in parking.conf for parkpos, I find that the first call is always parked on 701. Is this a bug?
Jim
> On Wed, 25 Feb 2004, [iso-8859-1] Øyvind Johnsen wrote:
>
>> I tried at first Theo's chan_sccp. (No Dialtone)... I have now tried the
>> lambda-solution and asterisk keeps crashing every time I have used the
>> 7912... I guess that skinny / asterisk have some way to go before it is
>> of
>> any u
Hi!
> Does anyone here have any experience with web based soft clients for *?
I played with two:
1. The FWD ActiveX works ok, but it is unsigned so that you'll have to
adjust your Internet Explorer security settings
http://fwd.pulver.com/callme.php?userid=yourFWDnumber
2. Xten's X-web (version
At 08:26 AM 2/25/2004, you wrote:
Ernest,
Wondering if I could could get some feedback about your system and how
it's
performing, as we are also considering replacing our existing pbx with *
...
How many phones do you have total using * .. 13 ? How many co lines ? pri
?
Are you using a long dista
Hi!
> >This may sound silly but how can I say to asterisk that new number have
> >been dialed and that it has to treat these as a new extension ?
> >
> >I mean: I have received a call, and now I want that asterisk execute the
> >command, by example call forwarding, recording... that I can do when
why not load a client on their system they are using? There are quite a
few iax soft phones for both linux/win32
On Wed, 2004-02-25 at 13:58, [EMAIL PROTECTED] wrote:
> You may be right here. I was thinking of an ActiveX plug-in. I don't expect them to
> use public internet kiosks so they should b
Found this link from a google search
http://www.dairiten.com/webiax/
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting [EMAIL PROTECTED]:
> You may be right here. I was thinking of an ActiveX plug-in. I don't expect
> them to use pu
My guess was the 100 presented too low an impedence to the line. So, I took
an answering machine that had a phone jack on it (pass-through). I plugged
the ans. machine into the filter and the 100 into the ans. machine.
Everything works now. I can also try a second filter.
Thanks,
Tom Schaefer
___
I went back on my old CDR's they are correctly recorded, so is this
feature introduced recently with a new version upgrade ?
>> instead of macro-dialout if I directly dialed through the [intern] I
get
>> the correct results. Some how asterisk think I dialed extension "s"
instead
>> of the numbe
I've got several of the Polycom IP 500's as well. I was using snom 200's.
They were horrible speakerphones and the echo inside the handset made it
almost unusable. The 100 was completely unusable. For less than $60 per
phone more, I got a phone that I have to spend les
You may be right here. I was thinking of an ActiveX plug-in. I don't expect them to
use public internet kiosks so they should be able to use the ActiveX approach. I was
hoping that something IAX based could be found as it would make the connectivity
easier and open port risk reduced.
Michael
We implemented 21 Polycom IP 500 SIP phones in December and the voice
quality with Ulaw is very good. We also use g729 and IAX to connect two
sites together on DSL and voice quality sounds pretty good as well. No
one has complained yet.
The speaker phone is pretty good, and occasionally on POTS
Want to add a little something about the BG speakerphone. It is not very loud,
but it is pretty good other than that. I have tested snom 100, snom 200, and
siptones and BG is the only one that was remotely close to usable. The bg
actually has a full duplex speakerphone which is not all the commo
Hi!
> instead of macro-dialout if I directly dialed through the [intern] I get
> the correct results. Some how asterisk think I dialed extension "s" instead
> of the number I dialed.
Yes, that's unfortunately the way it works with macros. Either don't use
a macro, or make sure that after the m
On Wednesday 25 February 2004 08:26 am, Peer Oliver schmidt wrote:
> Good day,
>
> I am in the middle of getting my self some hard phones. Anyone care
> to comment on the *voice* quality of the following phones:
>
> Cisco 7960
> Siptone II
> SNOM
> Budgetone
>
> I have seen a few reviews, but none
Problem seems to be due to the fact that I use a macro and it has a "s"
as the matching extension, but if the Dial command dial out correctly
why the CDR not get recorded correctly. As for testing, I directly
dialed out without the macro command, and it works fine. But my
extensions.conf is based
By web based do you simply mean a UA that is *deployed* using "the web"
(http) or do you also mean that to include tunneling of media over 80/443?
Any Java based softphone could easily be turned into an applet, thus
satisfying the web-based part of your query.
An Active X component is nothing more
Chris Clifton wrote:
We also lost connectivity to them for approx 1.5 hours this morning.
Send a traceroute of the outage to [EMAIL PROTECTED] as we were not down.
Jeremy McNamara
- Original Message -
From: "Joseph Finley" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wed
I think xten is supposed to have an active X control version of their
softphone that would probably do what you are talking about.
On Wed, 25 Feb
2004, Michael Graves wrote:
> Hello All,
>
> Does anyone here have any experience with web based soft clients for *?
> I'm thinking about putting a p
Hi all.
Just got 10 aastra 390s and searching for some page ort resources with
ADSI programing guide/examples.
P.S. These phones rocks :)
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBS
Title: Unreliable Fax
The Asterisk system only has Digium FXS and FXO boards, and the auto attendant routes fax calls to one of the FXS boards. It works sometimes, but most long faxes stop in the middle and some faxes do not appear to even be answered. It may depend on which of the two FXO
At 09:21 AM 2/25/2004, you wrote:
Ernest W. Lessenger wrote:
> At 08:15 AM 2/25/2004, you wrote:
>
>> This may sound silly but how can I say to asterisk that new number have
>> been dialed and that it has to treat these as a new extension ?
>>
>> I mean: I have received a call, and now I want that
On Wed, Feb 25, 2004 at 03:22:30PM +, Fran Boon wrote:
> >>When people talk about voicemail these days, they mean 'voicemail2'
> >Really? In the docs somewhere, it shows voicemail2 as deprecated.
>
> ok, 'voicemail2' has been renamed as 'voicemail'
Ah. Non-obvious. Now I'll have to go back an
Ernest W. Lessenger wrote:
At 08:15 AM 2/25/2004, you wrote:
This may sound silly but how can I say to asterisk that new number have
been dialed and that it has to treat these as a new extension ?
I mean: I have received a call, and now I want that asterisk execute the
command, by example call fo
Greg,
When using your patch, when the file is being played... The MOH stops
for the caller while that sound file is played to me. Then the music on
hold continues and if I hit # the music on hold stops once again and
let's me talk.
I wonder if we can stop that from happening?
Sincerely,
Matt
On Wed, 2004-02-25 at 17:30, Michael Graves wrote:
> Hello All,
>
> Does anyone here have any experience with web based soft clients for *?
> I'm thinking about putting a page up on our corp web server that would
> let staff in the field connect to our in-house phone system via the
> internet. Thi
I'd reach for the Oxometer on that one - 36k shouldn't make any difference.
However, the X100P may be introducing some capacitance on the line that
would affect the ADSL signals - but the purpose of filters is to stop this
problem. Maybe it's worth trying another filter between the X100P and yo
Hello All,
Does anyone here have any experience with web based soft clients for *?
I'm thinking about putting a page up on our corp web server that would
let staff in the field connect to our in-house phone system via the
internet. This could help staff making overseas calls while on trips,
withou
At 08:15 AM 2/25/2004, you wrote:
This may sound silly but how can I say to asterisk that new number have
been dialed and that it has to treat these as a new extension ?
I mean: I have received a call, and now I want that asterisk execute the
command, by example call forwarding, recording... that I
We also lost connectivity to them for approx 1.5 hours this morning.
- Chris Clifton
- Original Message -
From: "Joseph Finley" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, February 25, 2004 10:52 AM
Subject: [Asterisk-Users] Sorry, OT (NuFone)
>
>
> Is anyone having pr
This may sound silly but how can I say to asterisk that new number have
been dialed and that it has to treat these as a new extension ?
I mean: I have received a call, and now I want that asterisk execute the
command, by example call forwarding, recording... that I can do when I
dial a precise
At 05:49 AM 2/25/2004, you wrote:
The Snom 200 phone mostly functions well, however the phone's logic is more
oriented to european telephony and several of the functions do not work in
a manner that one might consider 'standard' in the US. It's light-weight,
pulls across the desk when the handset c
At 05:26 AM 2/25/2004, you wrote:
I am in the middle of getting my self some hard phones. Anyone care to
comment on the *voice* quality of the following phones:
Cisco 7960
Siptone II
SNOM
Budgetone
I have seen a few reviews, but none go to deep into the voice quality
issue.
I have not received an
Joseph Finley wrote:
Is anyone having problems registering with NuFone? My system has not been
able to register over the last couple hours. I've sent a support email in
without any answe as of yet.
Seems to be working just fine here.
B.
___
Asterisk-Us
Disregard, it's back up now.
Joe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph Finley
Sent: Wednesday, February 25, 2004 10:52 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sorry, OT (NuFone)
Is anyone having problems registering with
Is anyone having problems registering with NuFone? My system has not been
able to register over the last couple hours. I've sent a support email in
without any answe as of yet.
Regards,
Joe
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://li
That's great, but the Polycom phones and the Cisco phones are already
capable of this. We just need asterisk to comply:
>From the Polycom admin manual at:
http://www.polycom.com/common/pw_item_show_doc/0,1276,2545,00.pdf
5.2.6 Shared Call Appearance Signaling
A shared line is an address of recor
Tim Sailer wrote:
one more thing which one is newer versionand has mysql support
voicemail or voicemail2
'voicemail' is deprecated.
When people talk about voicemail these days, they mean 'voicemail2'
Really? In the docs somewhere, it shows voicemail2 as deprecated.
ok, 'voicemail2' has been ren
Post your configs, maybe we can help. :)
Sincerely,
Matthew Marlowe
Gear 3 Technologies, LLC
609.252.1155 x614
www.gear3.com
(00)
>< Choose a job you love, and you will
/||\ never have to work a day in your life.
=/\=
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Comments are inline.
Robert
Jeroen Rikhof said:
> Hello,
>
> Can somebody give me some information about:
>
> 1. How stable Asterisk is?
My experience and from what I have read on the list is that it is very
stable if run on stable hardware and you don't mess with the program code.
If you mess wit
On Wed, 25 Feb 2004, Peer Oliver schmidt wrote:
> Greg Boehnlein wrote:
>
> > On Tue, 24 Feb 2004, Greg Boehnlein wrote:
> >
> >
> >>Hello all,
> >>I have an application where I am attempting to use Agents and
> >>CallQueues to distribute inbound calls to remote users on cell phones. The
> I am in the middle of getting my self some hard phones. Anyone care to
> comment on the *voice* quality of the following phones:
>
> Cisco 7960
> Siptone II
> SNOM
> Budgetone
>
> I have seen a few reviews, but none go to deep into the voice quality
> issue.
In theory (and mostly in practice
On Wed, 25 Feb 2004, Matthew B Marlowe wrote:
> I believe this is what you want something similar to:
>
> Queues.conf:
> Member => Agent/000
>
> Agents.conf:
>
> (I also added a patch from ?someone? To play a file before call is
> accepted)
That was me! ;)
> Agent => 000,,Test Agent (This a
Vic Cross wrote:
G'day Marc,
On Wed, 25 Feb 2004, Marc Fargas wrote:
I’m in trouble with SIP. I’ve got a SIP FXS gateway from www.micronet.info
(SP5002/S) and traed to register to asterisk, It seems to autentícate but
sniffing the net it shows a 407 proxy authen required error message and I
can
Jean-Denis,
Your suggestion works for me indeed. I had to test around a little, and
I had to put the entire MSN number in the extension field. How come you
only put part of it in there? Did you specify something about the common
part in the modem.conf file?
My idea with the DNID variable does not
On Wed, Feb 25, 2004 at 01:46:39PM +, Fran Boon wrote:
> >one more thing which one is newer versionand has mysql support
> >voicemail or voicemail2
>
> 'voicemail' is deprecated.
> When people talk about voicemail these days, they mean 'voicemail2'
Really? In the docs somewhere, it shows
On Wed, 25 Feb 2004, [iso-8859-1] Øyvind Johnsen wrote:
> I tried at first Theo's chan_sccp. (No Dialtone)... I have now tried the
> lambda-solution and asterisk keeps crashing every time I have used the
> 7912... I guess that skinny / asterisk have some way to go before it is of
> any use other t
atif wrote:
I need some tips on configuration of voicemail with mysql...
http://voip-info.org/tiki-index.php?page=Asterisk+voicemail+database
here is my voicemail.conf
**voicemail.conf***
[general]
dbhost=localhost
dbname=asteriskvmusers
dbuser=root
On Wed, 25 Feb 2004, Peer Oliver schmidt wrote:
> Budgetone
>
> I have seen a few reviews, but none go to deep into the voice quality
> issue.
I don't mind the voice quality, I just wish it would always be working
when I picked up the handset. Mine tend to lock up (they are behind a
firewall, I
> ?yvind Johnsen ([EMAIL PROTECTED]) wrote:
>> Hi.
>>
>> I have set up * in our lab here at work and got the 7940 up and running
>> OK. But the 7912 wont work. It registers sometimes. When I call it it
>> rings, but when I lift off the handset it just keeps ringing and the
>> call
>> is not set up.
G'day Marc,
On Wed, 25 Feb 2004, Marc Fargas wrote:
> Im in trouble with SIP. Ive got a SIP FXS gateway from www.micronet.info
> (SP5002/S) and traed to register to asterisk, It seems to autentícate but
> sniffing the net it shows a 407 proxy authen required error message and I
> cannot make an
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