On Thu, 2003-07-24 at 22:38, [EMAIL PROTECTED] wrote:
> I'm having the following problem:
>
> I call into my Asterisk box (RedHat Linux 9.0, 1 Digium X100P card)
> to access voicemail. After dialing the appropriate extension I get
> voicemail, am presented with the user's unavailable message, an
Try looking drunkencoder.com/asterisk
On Thu, 2003-07-24 at 22:16, Andy Hester wrote:
> I have searched and not located this patch...is there a specific place that
> I need to look, or a specific file name?
>
> Andy
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMA
hi guys,
have anybody tried using audiocodes sip fxs against
asterisk? how's the device fairing?
~kelvin
After speaking with Martin @ Digium, I have the following answers.
The driver Wct4xxp determines the number of channels by the signaling type set in the
/etc/zaptel.conf file.
For example if all the spans used b8zs,esf your spans would look like this:
Span 1 Zap/1 - Zap/24
Span 2 Zap/25 - Za
A call is placed via IAX2 from one asterisk to another, to a TDM400
channel whose extensions.conf entry is
exten => 502,1,Dial(${COLIN})
exten => 502,2,Congestion
If this channel is already busy when called, the call is instantly
hungup, without the caller hearing the congestion tone.
The log
Hello,
After further testing. I have to manually issue the reload command after
every call to avoid the 'Unable to allocate private structure' error.
Pretty bad :(
Foong
- Original Message -
From: "Chee Foong" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, July 25, 2003 10:37
I'm having the following problem:
I call into my Asterisk box (RedHat Linux 9.0, 1 Digium X100P card)
to access voicemail. After dialing the appropriate extension I get
voicemail, am presented with the user's unavailable message, and can
leave a message normally.
The problem comes when I press
I have searched and not located this patch...is there a specific place that
I need to look, or a specific file name?
Andy
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of John Todd
> Sent: Thursday, July 24, 2003 6:14 PM
> To: [EMAIL PROTECTED]
> Subje
Michael,
my mistake - more testing confirmed that the wrapper did not update in the
correct location. Asterisk was still using 0.5.3. Replaced with 0.5.4 and
the call is no longer dropped.
Asterisk still reports H.323 CONTROL PROTOCOL ERROR (Roundtrip Delay) but
it keeps hold of the call.
Hello,
I have a voip endpoint calling the asterisk, when this endpoint press
extension 2, asterisk will dial to another voip endpoint. However when
asterisk try to dial the second endpoint I go the following error:
Can somebody help me?
-- Executing Ringing("H323/ip$202.75.145.111:2131/15497", "
Hello.
Does asterisk detect fax tones so it can switch to an extension where a
regular fax machine is ?
I've looked at the archives, and there seems to be a lot of debate about the
fax issue in general.
The last one, in January, I think, talked about a user that had fax tones in
his voicemail,
Siggi Langauf wrote:
Are you running CCM 3.3(2), too?
No idea, I avoid dealing with CCM at all I fought tooth and nail to
stop them from wasting money on it, but they wouldn't listen to me.
OTOH, your scenario doesn't look like it's involving any Skinny phones, so
maybe your CCM doesn't s
Hi Michael,
I have just updated to 0.5.4 and the problem is still there. Are there any
parameters or logs that I should be checking?
When I run SJPhone (h323) direct to the Cisco 2600 fxo gateway - the call
remians up without error.
When I run SJPhone (h323) to Asterisk and then to a SIP e
http://www.automated.it/guidetoasterisk.htm
http://asterisk.gnuinter.net/
http://www.digium.com/index.php?menu=documentation
If you
need more ask jtodd in #asterisk IRC. He has some good example config
files.
Erik
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL P
yes, i agree, we never really felt the need to use unity, *'s vm is
functionally ok with callmanager
(except for the message waiting indication, or is there?) can *'s vm send a
MWI to the callmanager?
- Original Message -
From: "Siggi Langauf" <[EMAIL PROTECTED]>
To: "Asterisk user list"
Hi,
I see that there's been some very light discussion on having a standard time
and date stamp in VM. How can I implement it today? (About to offer a
system to a customer but they need the stamp to tell when people called.)
Thanks,
--
Steve
__
This sig is pendi
On Thu, 24 Jul 2003, Jeremy McNamara wrote:
> Siggi Langauf wrote:
>
> >We're running Cisco CallManager 3.3 with Cisco 7940 and 7960 Skinny
> >phones. * is registered to the CallManager as an H.323 gateway (using the
> >chan_oh323 driver, chan_h323 didn't work with the Cisco cluster).
> >
> >
>
>
Is there any kind of Configuration guide
available?
I've been trying to get a SIP Soft phone to work
and all I get is:
NOTICE[81926]: File chan_sip.c Line 4716
handle_request): Registration from ' failed for
'yyy.yyy.yyy.yyy'
I know I'm missing something in the configuration.
But don
Siggi Langauf wrote:
We're running Cisco CallManager 3.3 with Cisco 7940 and 7960 Skinny
phones. * is registered to the CallManager as an H.323 gateway (using the
chan_oh323 driver, chan_h323 didn't work with the Cisco cluster).
chan_h323 most certainly works with CCM. We have over 500 users u
I would like to see your code...
sounds great
Dave
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
On Wed, 23 Jul 2003, Troy Settle wrote:
> Funny. I just subscribed to this list to ask the exact same question.
>
> The application I have in mind though, would be a little more intense. What
> I would like to create, is a unified messaging center for voice, fax, and
> follow-me service (home, o
On Wed, 23 Jul 2003, Ronnie Earle wrote:
> I'm sure asterisk would make a great stand alone voice mail server.
> Basically I want to get rid of our voice mail system and replace it with
> *, but the problem is we use a cisco cluster with skinny clients. So I
> was thinking the way to contact a * s
Just curious if anyone has ran into this problem before, and if so if they
were able to work around it. When I dial into a MeetMe conference via an
IAX connection, my call is dropped if I enter an invalid meeting ID. I am
first played the greeting conf-invalid, then the call is dropped. If I
cal
Hi,
I see that there's been some very light discussion on having a standard time
and date stamp in VM. How can I implement it today? (About to offer a
system to a customer but they need the stamp to tell when people called.)
Thanks,
--
Steve
__
This sig is
Hi Pauline,
On Thu, 2003-07-24 at 22:21, Pauline Middelink wrote:
> > > The Gesko Ikarus 1200S analog telephone has advanced callerid
> > > capabilities. When used with an ATA186, it show the username
> > > and the phonenumber of the caller. (or whatever you let *
> > > tell it)
> > > http://www
Hi,
I want to utilize the above card to achieve the following
* Attach two ISDN connections (4 channels) to the outside world
* Have two internal ISDN connections (4 channels) to the inside
* Have multiple SIP phones connected
Anyone out there having success in utilizing the C4s ISDN ports to
pro
Turned out to be a incompatibility with my server hardware. This version of the
TDM400P was never going to work. So this server will work great as an VoIP only server.
-- Original Message --
From: "Stefano Finetti" <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTEC
On Thu, 24 Jul 2003 around 18:26:11 +0200, Armand A. Verstappen wrote:
> On Thu, 2003-07-24 at 17:02, Pauline Middelink wrote:
> > The Gesko Ikarus 1200S analog telephone has advanced callerid
> > capabilities. When used with an ATA186, it show the username
> > and the phonenumber of the caller. (o
> My Eicon ISDN card turned up today so - plugged it in and went through
> the modem.conf. It reports unable to open /dev/ttyI0
>
> The problem is I have never used ISDN with Linux - let alone a telephony
> app - and I have no idea even where to start. Some pointers would be
> appreciated.
Check o
You need to do something like
I do not have the Active-X IAX client so the above CLASSID is wrong. I am
sure the types are too.
Then to make some things work for need to do something like
Then make some Java script methods to call Active-X methods like
function myClickMeButton()
{
Just be careful if you want CallerID on the FXO side. Only the L2 FXO cards
with fairly new firmware will pass through the necessary audio.
-wade
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jerk Face
> Sent: Thursday, July 24,
Sure, you just have to have the mailbox keyword before you define each
"channel => a"
You should have a stutter tone & MWI
regards
Martin
On Thu, 24 Jul 2003, Jerk Face wrote:
> Does the T100P support message waiting on an Adtran TSU 600E with FXS cards
> installed?
> So basically:
>
> Asterisk
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Brian Capouch
> Sent: Thursday, July 24, 2003 1:02 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Changes to reset method for ATA186?
>
> I am trying to do a "factory reset" of a
Does the T100P support message waiting on an Adtran TSU 600E with FXS cards
installed?
So basically:
Asterisk w/T100P -> Adtran TSU 600E -> Analog phone
Will I be able to receive the stutter dial tone on the analog phone?
Thank you for your time
___
If you have a unit that is v2.16, it is possible that it has been
locked and is un-resettable.
See http://www.loligo.com/asterisk/cisco/ATA-186-guide.v20030628.txt
JT
At 1:01 PM -0500 7/24/03, Brian Capouch wrote:
I am trying to do a "factory reset" of an ATA186 using the
widely-available inst
Hello All,
I was just after a few pointers. I have just setup my Redhat 9 linux box
with asterisk. Internal SIP call working fine.
My Eicon ISDN card turned up today so - plugged it in and went through
the modem.conf. It reports unable to open /dev/ttyI0
The problem is I have never used ISD
I am trying to do a "factory reset" of an ATA186 using the
widely-available instructions (basically dialing "FACTRESET#" on the
keypad while at the menu prompt).
I have done this a number of times before with success, but on this unit
the lady spells out "P A S S W D" when I finish up the entry
Good day,
anyone had success finding a SIP or H.323 phone with intercom
capabilities? Softphone would be preferred?
TIA
rgds
pos
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Peer Oliver schmidt wrote:
Is there a Debian package available for asterisk-oh323, or the chan_h323?
No debian package for asterisk-oh323, but it's straightforward
how to build it from source.
http://www.inaccessnetworks.com/projects/asterisk-oh323
If yes, where might I find one?
Thanks
rgds
pos
Is there a Debian package available for asterisk-oh323, or the chan_h323?
If yes, where might I find one?
Thanks
rgds
pos
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Yes, * would work well with a PRI. You should only need one number
assigned, but then use the ANI (or DNIS; depends on your access
method) to send them to the right mailbox.
I would suggest a 2x2ghz box with 512m to handle 4 PRIs for voicemail
only. That might be slight overkill, but better t
There was a thread on FWD failures yesterday and indeed it didn't work for
me at 9:00 in the morning but by 10:30 all was fine - I'd made no changes
to *. It looks as though there's some tinkering going on at the FWD end.
Iain
--On Thursday, July 24, 2003 12:32:00 -0400 Leif Madsen
<[EMAIL
Hi,
I have a cheap GE cordless phone (ES26732) with something named Dual Caller
ID.
It displays both name and number with Asterisk and the default setting of
ATA186.
It cost about 45 EURO.
BR,
Dan
- Original Message -
From: "Armand A. Verstappen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTEC
On Wed, 23 Jul 2003, Yifang Dai wrote:
> I wish! My company just spend a lot $$ on the shinny CCM phone system, so I
> don't think I can change that easily... But if I can get asterisk to
> talk to CCM via h323, and prove it's usefulness, I might have a chance
> to use * in the branches...
Well,
I'm wondering if anyone else has gotten something similer to this? I
had FWD working fine on the asterisk box, then all of a sudden it just
stopped working. I get the following errors (just keeps looping)
*CLI> DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping
retransmission on
On Thu, 2003-07-24 at 17:02, Pauline Middelink wrote:
> The Gesko Ikarus 1200S analog telephone has advanced callerid
> capabilities. When used with an ATA186, it show the username
> and the phonenumber of the caller. (or whatever you let *
> tell it)
> http://www.gesko.be/idgg004.htm
>
> Pr
I'm new to the Asterisk software but have successfully set it up to make
and receive calls using FXO cards, voicemail transfer etc.
I can successfully call the Digium test IAX using the examples provided.
I have signed up for an IAX tel account and got a number.
The extensions have been set up as
Brad's recent list of enhancements look good, but I haven't looked
at the code yet. If the code looks good, I hope it will be committed
to the project CVS.
Here's a partial list of enhancements that I would like to see in
Comedian Mail. I am probably interested in helping to fund the
enhancement
To whom it might concern,
The Gesko Ikarus 1200S analog telephone has advanced callerid
capabilities. When used with an ATA186, it show the username
and the phonenumber of the caller. (or whatever you let *
tell it)
http://www.gesko.be/idgg004.htm
Price is 77 euro something and available
Kim C. Callis wrote:
> I was thinking of adding QoS to my Linux based router. I thought I would
> add all my IP phones and my * box into a VLAN, and then would do a QoS
> setup for that particular VLAN. Has anyone did any QoS setups for better
> performance? Has it made any change to the performan
Steven Thomas wrote:
Hi,
I'm using a Cisco 7960 with a SIP load, and a Cisco 2600 router with an FXO
port. Asterisk talks to the router via h323 and opens a call and connects
with no problem.
At exactly 74 secs (timer on the phone) the call drops, and Asterisks
displays this message:
-- H32
On Thu, 2003-07-24 at 00:26, Dave Cotton wrote:
> Which language are to translating into?
>
> I am about to start on French, pound = # = dièse
> Maybe one day someone will translate into English :)
> pound = # = hash.
> --
> Dave Cotton <[EMAIL PROTECTED]>
Sorry for rambling off-topic but I c
Some more unusual ones:
http://www.muppetlabs.com/~breadbox/intercal-man/tonsila.html
-Original Message-
From: Gary Gapinski [mailto:[EMAIL PROTECTED]
Sent: 24 July 2003 14:37
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] the 'pound' and '#' are the same? (OT
Rambling)
On Thursda
On Thursday 24 July 2003 01:21, John Laur wrote:
> I haven't ever found any really interesting technical terms for @, %,
> or ?. I'd be interested in hearing some..
http://www.unicode.org/charts/PDF/U.pdf, as well as the rest of the
Unicode site, is a good place to start. Alas, the characters
On Thu, 2003-07-24 at 07:03, Cerrajetto wrote:
> Hello!
>
> Is there a way to communicate from Asterisk to a TTS server?
>
> I've seen festival.conf, but it seems that it works only with Festival server.
>
> Thank you.
And whats wrong with that?
--
Steven Critchfield <[EMAIL PROTECTED]>
_
On Thu, 2003-07-24 at 04:46, John WALTER wrote:
> Hi
>
> I have some trouble getting asterisk to compile on my system. I get
> unresolved external symbol in enum.c et srv.c on res_ninit, res_nsearch
> and res_nquery. I've looked through my /usr/include/resolv.h file, and
> endeed I didn't found an
Return Receipt
Your Re: [Asterisk] help with extension switching
document
:
Hello!
Is there a way to communicate from Asterisk to a TTS server?
I've seen festival.conf, but it seems that it works only with Festival server.
Thank you.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/a
Steve,
May be entirely not related depending on your situation, but here goes.
I had some problems using gnuGK with Microtelco, because gnuGK didn't
send an unsolicited status message periodically. Because the Microtelco
end was using this as a 'heartbeat', it would terminate the connection,
alw
great. I noticed that the patch has been added to the main cvs.
matteo.
Il gio, 2003-07-24 alle 00:51, Ryan Tucker ha scritto:
> On 23 Jul 2003 20:38:42 +0200, Brancaleoni Matteo <[EMAIL PROTECTED]>
> wrote:
> > here's the patch that fixes the problem with dtmf INFO
> > messages, when pressing *
> > "Ryan" == Ryan Tucker <[EMAIL PROTECTED]> writes:
>
> Ryan> They are the same key. I'm not sure how the # came to be associated
> Ryan> with the word "pound", but in American English at least, they're the
> Ryan> same key.
>
> The weight measurement pound is abbreviated lb. # looks simi
> "Ryan" == Ryan Tucker <[EMAIL PROTECTED]> writes:
Ryan> They are the same key. I'm not sure how the # came to be associated
Ryan> with the word "pound", but in American English at least, they're the
Ryan> same key.
The weight measurement pound is abbreviated lb. # looks similar in
some ha
Hi Alex,
About:
> WARNING[12301]: File chan_zap.c, Line 5342 (zt_pri_error): PRI: Read
on 203 failed: Unknown error 500
Similar warnings on my system are generated due to shared IRQ t1xxp and
ide0 (/proc/interrupts).
Just my 0.02$
Konrad
-Original Message-
From: [EMAIL PROTECTED]
[mail
Hi
I have some trouble getting asterisk to compile on my system. I get
unresolved external symbol in enum.c et srv.c on res_ninit, res_nsearch
and res_nquery. I've looked through my /usr/include/resolv.h file, and
endeed I didn't found any declaration of such functions. It seems to be
some res_ini
>
> It's quite curios that a so little sign have so many names ;-)
> --
Yep. In my neighborhood in Brazil, the guys call it "lasagna"..
yes... that italian wonder...
Because it's look like a lasagna...hhehe
Isamar
___
Asterisk-Users mailing list
[EMAI
On Thu, 2003-07-24 at 10:12, Stefano Finetti wrote:
> Just to say, in order to continue this interesting digression and to talk
> about * globalization ;)
Globali{s|z}ation, now we"re talking, http://thevoice.digium.com is
great for an American, could we not have various thevoices, I'm sure
every
From: "Brad Bergman" <[EMAIL PROTECTED]>
>
> Curiously also, to digress, Nortel in Meridian Mail and its derivatives
> favours the term "number sign".
Just to say, in order to continue this interesting digression and to talk
about * globalization ;)
In italy the # sign is often called "Cancellet
From: "Matthew Pallotta" <[EMAIL PROTECTED]>
> Is there any way to run asterisk without a fxo card? I am looking only
> run SIP and a single fxs card.
>
Sure there is.
You can easily set up a pure VoIP box (is the easiest configuration of *)
and use for "internal" comm environment (obviously,
On 24 Jul 2003, Dave Cotton wrote:
> On Thu, 2003-07-24 at 20:15, johncn wrote:
> > the POUND and # are the same key in the telephone's keypad. If they
> > are same, how could we understand the following message:
During playback of a voicemail message, # fast forwards; any other time, #
exits vo
Hello all,
Can I use asterisk to call party A then call party B and finally connect
party A to party B, so they can talk to each other?
Is this possible?
Thanks
Foong
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/l
70 matches
Mail list logo
