Hi guys,
I've been searching for a R2MFC E1 card that works with asterisk. As far as
i could find, i got the Dialogic DTI/301SC card. Is there a way to make it
work with asterisk or anyone else can recomend me another brand that
actually works with asterisk?
Regards
-=Raul=-
Dinesh Nair wrote:
hey * folk,
am trying to configure a WellGate 3504A FXS SIP ATA
(http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set
up two SIP clients in sip.conf as follows:
The Unathorized really suggests that the password is wrong for 1235.
Check that you have the same
Dinesh Nair wrote:
Proxy-Authenticate: Digest realm=asterisk, nonce=514a6d7a
Don't forget to change your realm to something that is unique for your setup...
realm= in sip.conf :-)
Regards,
/Olle
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Hi everyone!
Is there a way to let Asterisk connect to/interface with the Micronet
SP5210 SIP server ( http://www.micronet.info/Products/voip/SP5210.asp )?
It does not support IAX, but maybe there is another way...?
Greetings,
Evert Meulie
___
On Wed, 2004-09-08 at 09:43, Vladyslav wrote:
HI!
Have a weird problem with X-lite Meetme.
When X-Lite user are join to conference room NOT first one, than
X-Lite user do not hear anything. This problem gone when X-Lite user get
into conference room first (when nobody there).
sip.conf
I am attempting to transfer a number that comes in on an FXO port back
out the same port. The service has 3 way calling and transfer and these
options are specified in zapata.conf . Some config...
zapata.conf
[channels]
;
context=incoming
signalling=fxs_ks
echocancel=yes
On Wed, 2004-09-08 at 04:43, HengWee Chin wrote:
Hi all,
I have the following setup
PSTN - ASTERISK - IVR (using dialogic card)
1) Caller id information is presented to asterisk during the first and
second ring.
2) Hence, Asterisk waits for 2 rings before pickup the call and
Hello Nicols,
On Wed, 2004-09-08 at 14:17, Nicols Gudio wrote:
Did you try asterisk manager? You can execute all of the cli commands
and much more. Just enable it in /etc/asterisk/manager.conf and read
manager.txt in the asterisk docs directory.
No, I haven't tried asterisk manager. To be
Hi,
-Original Message-
I get an Intertex IX66 and I'm trying to connect my * behind
this SIP router. I can register my Polycom phones on * but
the sound on the phones is just one way.
Someone fight with the same problem with this router?
Can you tell us how you configured the
On Thu, 2004-09-09 at 09:47, Umar Sear wrote:
On Wed, 2004-09-08 at 09:43, Vladyslav wrote:
HI!
Have a weird problem with X-lite Meetme.
When X-Lite user are join to conference room NOT first one, than
X-Lite user do not hear anything. This problem gone when X-Lite user get
into
Michael George wrote:
To follow up on this, I heard back from Digium and they asked the
configuration of my TDM. It was: FXO,FXS,FXS,FXS. They said they have had
report of this configuration being a problem and that I should change it to
FXS,FXS,FXS,FXO.
Before the change the system would
Hello list
When i'm trying to dial into our pstn the following errors occure:
-- Executing Dial(SIP/snomsip-dbd0, /2100) in new stack Sep 9 10:02:22
WARNING[59409]: channel.c:1901 ast_request: No channel type registered for
''
Sep 9 10:02:22 NOTICE[59409]: app_dial.c:715 dial_exec: Unable to
Asterisk is up running.
SIP phones (SJPhone, xlite i.e) can call an
extension and get a voice message...
..then expect the customer to push 1, 2 or 3 to be
sent to other extensions.
This works for SIP phones.But when mobiles or ordinary phones
(what is the tech word for this?) call the *
On 09/09/2004 14:12 Olle E. Johansson said the following:
The Unathorized really suggests that the password is wrong for 1235.
Check that you have the same secret on both asterisk and the welltech.
that's the first thing i looked for, and i've more than triple checked
this. the passwords are
bri-stuff.0.1.0-RC4a on debian sid with 2.6.7 kernel
isdn sk:
Bus 0, device 18, function 0:
Network controller: Digi International Datafire Micro V (Europe) (rev 2).
IRQ 17.
Master Capable. Latency=16. Max Lat=16.
I/O at 0xe400 [0xe407].
Non-prefetchable 32 bit
Hi Everyone
I have done the following steps with asterisk are as follows:
1. I have installed latest version of asterisk from cvs.
2. I have used database as postgresql.
3. I have installed Asterisk Prepaid 0.3.1 from
http://sourceforge.net/projects/asteriskbilling/
4. Now the asterisk calling
Araba, Michael wrote:
I HAVE simalr setup
PRI from Central OFFICE -- T100P--Asterisk---T100P--Panasonic PBX 576
I keep getting Alarms on the T100P interfacing the PBX. Do you have any
suggestions? I can't make calls across into the PBX.
I don't know about the Merlin Legend, but the Panasonic 576
Hi everyone!
Does anybody have russian (or/and ukrainian) language sound file
collection for Asterisk? (at least those required for the voicemail
system). Or maybe anyone could point me to the location where to get ones?
TIA
___
Asterisk-Users mailing
Vladyslav [EMAIL PROTECTED] writes:
Try disallowing GSM for [104]
It's disallowed in dialplan already.
It's commented out ;allow=gsm
Yes, but have you restarted Asterisk after commenting it out? Just
doing a sip reload isn't enough. Ask my Grandstream how I know... :-)
-tih
--
Tom Ivar
Hi Umar,
unfortunately I have not found a solution for my problem. I do not think
that there is any problem in the dial plan. The IVR that I have is not done
using asterisk. It is another application running on another machine with a
telephony card (dialogic).
From my understanding, the
hi list,
i want to use the astersik in conjunction with
the ser
so i followed the instructions provided on the
voip-info.org site
but when calling from one user to another it gives me
problem in the asterisk cli that
failed user authentication
my aim of doing this is to use the
On Thu, 2004-09-09 at 21:38, HengWee Chin wrote:
Hi Umar,
unfortunately I have not found a solution for my problem. I do not think
that there is any problem in the dial plan. The IVR that I have is not done
using asterisk. It is another application running on another machine with a
Olle E. Johansson wrote:
Dinesh Nair wrote:
This is just a quick note: The Wellgate's only register one user/password.
If they have 2 ports they register them as one. They don't allow more then
one password or username to be sent. They know about this problem but as far
as I know they have not
Brent D. Franks wrote:
Hello,
I recently upgraded to Sip 1.3.1 and noticed that the Reject Button is
no longer appearent on the screen when a second incoming call comes in
unless I press the hold button on the first call.
Does anyone have a work around for this to reject a call while
Sounds like you need to talk to polycom about a reduction in the
capabilities of thier phone after the upgrade and have them move the menu
option back..
- Original Message -
From: Tor Setane [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Thursday, September 09,
Hi everyone!
situation:
Asterisk-server A: 192.168.11.6
Asterisk-server B: 192.168.2.44
server B contains a register = username:[EMAIL PROTECTED]
But... when I boot it, I get:
Registered to '192.168.11.6', who sees us as 10.138.3.2:4569
Why doesn't server A see server B as 192.168.2.44??
All other
Hi,
I try to dial out through a Provider, but for that i need to be authenticated - it actually does not work !.
What kind of dial command do i have to use when i:
- Want to authenticate the user
- And dial with that authenticated user an extension
What i tried so far
exten =
On Wed, Sep 08, 2004 at 12:27:05PM +0200, Roger Schreiter wrote:
Within the bios menu I can't find any appropriate mean.
It depends on the board/bios.
Thanks for any hints!
Technik und Know How zu IRQs, Sharing, APIC und INT-Leitungen:
http://www.hardtecs4u.com/reviews/2002/irq/
This might
I am using asterisk in a small call center where agents not only receive
calls but also call outside themselves. The queues work fine. However,
when an agent has called outside and is already on the phone, then also
he gets ACD calls. Is there a way to stop ACD calls to agents who are
already off
I see your problem, unless you point out this is already the case:
Matthias Leeb wrote:
Hello list
When i'm trying to dial into our pstn the following errors occure:
-- Executing Dial(SIP/snomsip-dbd0, /2100) in new stack Sep 9 10:02:22
WARNING[59409]: channel.c:1901 ast_request: No channel type
Hi all,
due to the rather big email traffic regarding this issue, I decided to
publish the script so people can download it at their own risk... :-)
Please, visit:
http://www.leals.com/~mm/asterisk
for further information.
Regards,
Martin
___
Stuart,
I have 3 of the Polycom IP 300 SIP's. The first two I bought were not sip and I had
to load the SIP firmware and application on them. That was somewhat involved, but not
too difficult. The third phone I bought with SIP on it.
I plugged it in, configured it, and it works great. I
On Thu, 2004-09-09 at 06:38, HengWee Chin wrote:
I am wondering if there is any way or settings I can set to allow the caller
id to pass thro' asterisk and let the IVR pickup the caller id information.
This means that asterisk do not wait for 2 rings to process the call. Any
ideas?
Easy.
Hi,
We've got endpoints and gateways who have T.38 fax support. We
now use SER and Asterisk to do our routing and other
functionality, but fax doesn't seem to work. Asterisk complains
like this:
Sep 9 09:25:45 WARNING[467828746]: RTP Read too short
Sep 9 09:25:45 WARNING[467828746]: Asked to
Andreas Sikkema wrote:
Hi,
We've got endpoints and gateways who have T.38 fax support. We
now use SER and Asterisk to do our routing and other
functionality, but fax doesn't seem to work. Asterisk complains
like this:
Sep 9 09:25:45 WARNING[467828746]: RTP Read too short
Sep 9 09:25:45
On Thu, 09 Sep 2004 11:51:09 +1000, Jamie Carl [EMAIL PROTECTED] wrote:
Hey all,
Just a quick question. Are there any known issues using the zaptel
drivers on different linux distros? ie: is a 2.4 kernel the only
requirement?
I did successfully set up my X100P in 2.4 and 2.6 kernels.
Jamie Carl [EMAIL PROTECTED] wrote:
Just a quick question. Are there any known issues using the zaptel
drivers on different linux distros? ie: is a 2.4 kernel the only
requirement?
That should read 2.4 or later. I'm using the Linux 2.6.8 kernel
with the Gentoo distro. In theory, the
On Fri, 3 Sep 2004 20:13:45 -0300, Marconi Rivello
[EMAIL PROTECTED] wrote:
Hi,
How do I put a call on hold? If i press # the music on hold plays to
the other person, but asterisk asks for a number to transfer... I
don't want to transfer, I simply want to put the person on hold, so
he/she
Last night I updated to a custom 2.4.27 kernel, I also upgraded
asterisk. This morning I discovered Asterisk is no longer playing sounds
to users. ie when they go to the voicemail, asterisk says it's playing
vm-login but the user never hears it. It's not a firewall issue or
anything like this,
I believe this has something to do with the converter. With my
Sipura-2000 if I hit flash, it puts the person on hold and I get a new
dialtone to place a call. From there I can call another number, and if I
hit flash again, it 3 way calls them. If I hang up, it leaves the other
two people
On Thu, 9 Sep 2004, Johannes Hollerer wrote:
I try to dial out through a Provider, but for that i need to be
authenticated - it actually does not work !.
For my tests I did not need to be authenticated. This is what I used in
asterisk:
exten =
Do you know where it got the 10.138.3.2 IP from? Is it configured
anywhere on the server? Do you have
externip defined in that config file?
Evert Meulie wrote:
Hi everyone!
situation:
Asterisk-server A: 192.168.11.6
Asterisk-server B: 192.168.2.44
server B contains a register = username:[EMAIL
But the provider also has a gateway to provide the possibility to call to the pstn (and the pstn number exists) - so what i tried to achive is to call an external pstn number thru that gateway.
This works if i connect the xlite client directly to the provider - then i can dial the external
But the provider also has a gateway to provide the possibility to
call to the pstn (and the pstn number exists) - so what i tried to
achive is to call an external pstn number thru that gateway. This
works if i connect the xlite client directly to the provider - then
i can
Hi to all,
we have a community of people on an * box that use SIP softphones to talk each
other. Can you suggest me the quickest and simple way to let someone know who
is online without have to call one by one the persons to look if they are
present or not?? Something the user list in Microsoft
Hello Ariel's,
i got this back from welltech
For the 38 unit
It couldn't support only one account for the registeration as so far...
their reponse
We have the plan for this function...but it will be ok before the end of
Q4
--
Best regards,
Danny
Hello all,
I updated from CVS a few days ago and noticed that my calls just cut
out without reason.
The CLI says this: -- Hungup 'Zap/3-1'
It occurs without error or warning.
Is their a bug in CVS asterisk or libpri? This never occurred before.
We upgraded to 1.0 RC2 on Tuesday and
On Thu, Sep 09, 2004 at 07:53:22PM +1200, Richard Scobie wrote:
Michael George wrote:
To follow up on this, I heard back from Digium and they asked the
configuration of my TDM. It was: FXO,FXS,FXS,FXS. They said they have
had
report of this configuration being a problem and that I
I just downloaded it now off the CVS and it will no longer compile
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Has there been any development with call back transfers?
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Hi,
I have Asterisk w/ 192.168.1.1 and I setup IX66 to be 192.168.1.2 (I'm using
pppoe client and dyndns.org on IX66)
I setup on Local DNS Server my * box and after that I was able to register
my phones from the Internet.
I cannot understand my problem with one way sound... what is wrong on my
On Thu, Sep 09, 2004 at 11:19:02AM -0400, Deon Rodden said:
Marconi Rivello wrote:
On Fri, 3 Sep 2004 20:13:45 -0300, Marconi Rivello
[EMAIL PROTECTED] wrote:
How do I put a call on hold? If i press # the music on hold plays to
the other person, but asterisk asks for a number to
I was wondering if anyone knew how to do the following
Call comes in, gets put into a Queue, say `Sales`. Then the queue member
is presented with the option to exit the queue, yet have the phone
system sit in their place for them. When the virtual member reaches the
front, call back the
Anyone using the recently MAC OS X ? Version of asterisk ?
Thanks,
Francisco Perez-Landaeta
From: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Fri, 27 Aug 2004 13:08:24 -0500 (CDT)
To: [EMAIL PROTECTED]
Subject: Asterisk-Users Digest, Vol 1, Issue 5082
Send Asterisk-Users mailing
On Wed, 8 Sep 2004, JP Hindin wrote:
I have a bit of a conundrum, and I can't tell if Asterisk is doing
something daft, or whether I'm clean missing out why it's doing what it's
doing. So, I have a dialplan that looks a little like this:
[start]
include = dids
include
Hi
Il gio, 2004-09-09 alle 18:18, Michael Workman ha scritto:
I just downloaded it now off the CVS and it will no longer compile
this kind of messages are only waste on bandwidth space.
please:
* don't send a message like this
OR
* paste the error into the email, if you need support
OR
* try
Has anyone else successfully got such an arrangement to work, or is
there any plans to make the QSIG messages more parsable (maybe exposed
as variables in the dialplan)? Or at least not have the name overflow
the number? :)
QSIG passes callername and other variables by a mechanism that
Well this is what I am getting
[EMAIL PROTECTED] asterisk-addons]$ make
./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c`
cdr_addon_mysql.c:17:29: asterisk/config.h: No such file or directory
cdr_addon_mysql.c:18:30: asterisk/options.h: No such file or directory
Hi Everyone,
I've been asked to determine which phones our organization should go
with. And I've narrowed it down to the Polycom IP500 or the Cisco 7940.
From my travels through google, it's hard to find a definitive
comparison of the two phones. So I thought I would ask the people that
have
At 09:54 PM 9/7/2004 +0900, Benjamin on Asterisk Mailing Lists
[EMAIL PROTECTED] wrote:
For travelling, no SIP based device will be configure and forget.
Perhaps if you travel only within the US, you may be lucky most of the
time but pretty much anywhere else where IP addresses are scarce you
Can anyone on this list provide origination
from Aruba to the US or Canada?
Thanks,
--
Richard Cook
[EMAIL PROTECTED]
Tel: 705-497-9320- ext
2010
Blank Bkgrd.gif___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I found some postings from Google (notably from Mark Spencer) about
successful integration of a legacy Toshiba Strata system and Asterisk.
I am also facing that current dilemma. The general legacy solutions that
I can come up with is very easy -- either making Asterisk a proxy (or
frontdoor) to
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kenneth Shaw
Sent: Thursday, September 09, 2004 12:56 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Legacy Toshiba Phones
I found some postings from Google (notably from Mark
seems that asterisk isn't installed
Il gio, 2004-09-09 alle 18:48, Michael Workman ha scritto:
Well this is what I am getting
[EMAIL PROTECTED] asterisk-addons]$ make
./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c`
cdr_addon_mysql.c:17:29: asterisk/config.h:
Sorry for the OT, but I was simply wanting some rough guidelines from
someone who has bought one.
I resent having to fill out a form (every damn polycom partner I've found
in the uk will not publish a price) just to find out how much. If I have to
ask, is it too expensive ;)
And, from where
I plugged it in, configured it, and it works great. I really like the
polycom phones. They have a superb speakerphone.
(you can hear quiet whispers and people tapping pens on the desk.)
Just a note here... the IP300 doesn't have a mic on the speakerphone, it's
listen only.
On Thu, 09 Sep 2004 12:52:56 -0400, John Kington [EMAIL PROTECTED] wrote:
What about sip softphones that use STUN? I am especially interested in UK
because my daughter is going to study in London.
If she is going to be on a residential ADSL, that shouldn't be a
problem. I have friends in the UK
Does anyone know how to do this with the OH323 channel driver?
I want the local (7 digit dialing) to go out an h323 that I have registered
to a gatekeeper...
can I do something like
exten = _7.,2,Dial(OH323/ipofgatekeeper)
-Original Message-
From: Begumisa Gerald M [mailto:[EMAIL
Are the FXOs on the 2x on ports 1-2 or 3-4? Maybe it has to
do with *any* FXO on port 1...
Please get back with the list with your findings.
My experience led to a replacement from Digium, but the card is a
TDM400P with 4 FXO...now that I think of it, during troubleshooting
there
Due to the format of the message coming from the H323 channels included w/
Asterisk we were unable to use our gatekeeper.
For a quick solution we tried the OH323 channel drivers and can receive
inbound calls from the parent gatekeeper.
We are trying to do a dial to gatekeeper...
I am trying
exten
Hey all,
Did I see something on here about using an AGI script to do reverse
lookups via anywho.com? I have a PRI that only gets caller-id number and
no Alpha.
TIA,
--
Daniel Jimenez djimenez[at]pobox[dot]com
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[EMAIL
Hello,
Does anyone know how to
conference a call on the SNOM 200 phone? Whenever I push the cnf/tran button it
just hangs up on the active call. The manual says you have to push the cnf
function key but it doesn't appear in the lcd on my phone.
Thanks
-Matt
The Polycom IP500s do support customized ringtones and can use a customized
ALERT_INFO for all of them. One thing that is worth noting in this
comparison is that the IP500 doesn't support the XHTML microbrowser that the
IP600 does. Since they both use the same SIP application I am hoping they
Huddleston, Robert wrote:
Due to the format of the message coming from the H323 channels included w/
Asterisk we were unable to use our gatekeeper.
For a quick solution we tried the OH323 channel drivers and can receive
inbound calls from the parent gatekeeper.
We are trying to do a dial to
can you post the information on how you got that thing working?
thanks
hank
- Original Message -
From: Chris HARIGA [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Wednesday, September 08, 2004 8:55 PM
Subject: RE: [Asterisk-Users]
-I../asterisk seems to be the key
cd /usr/src
cvs checkout asterisk asterisk-addons
cd asterisk-addons
make
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brancaleoni Matteo
Sent: Thursday, September 09, 2004 12:05 PM
To:
Okay - read it... my configuration works... what I want
exten = XX,1,Wait,2
exten = XX,2,Dial(OH323/XX)
I want it to pass the 10 digits to the DIAL string... I'm not sure I
understand the macros
can I just put the ${EXTEN} in there??
-Original Message-
From:
- Original Message -
Hey all,
Did I see something on here about using an AGI script to do reverse
lookups via anywho.com? I have a PRI that only gets caller-id number and
no Alpha.
I have thought about this as well. Should be totally possible but first off
you could just change
what is the price range in us dollars?
- Original Message -
From: Jody N. Rudolph [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, September 09, 2004 11:32 AM
Subject: RE: [Asterisk-Users] Polycom IP500 vs Cisco 7940
The
*** Astricon 2004: Over 250 Asterisk professionals!
Astricon, the first Asterisk user's and developer's conference
is a success and we now have over 250 people registred.
Thank you for all your support of this event and please have
patience with us as we're trying to handle all details
with
Hello--
In the interests of playing around and wasting time, I've installed the latest version of the
Festival stuff, 1.95beta.
And, in the interests of future Asterisk-Festival connectivity, I applied the 1.4.3 patch to put in the
asterisk related routines. I did it by hand, but, it looks
On Sep 9, 2004, at 8:53 AM, Marcello Lupo wrote:
we have a community of people on an * box that use SIP softphones to
talk each
other. Can you suggest me the quickest and simple way to let someone
know who
is online without have to call one by one the persons to look if they
are
present or
Huddleston, Robert wrote:
Okay - read it... my configuration works... what I want
exten = XX,1,Wait,2
exten = XX,2,Dial(OH323/XX)
I want it to pass the 10 digits to the DIAL string... I'm not sure I
understand the macros
can I just put the ${EXTEN} in there??
Of course. The
Hi,
The big problem was with Ethernet name on bridge. U can find the real name
of your Ethernet on Cooperative Linux Console. Take a look @
http://www.techselesta.com/astwind.jpg and U will see a printscreen of my
error.
I have a Intel(R) PRO/100+ Alert on LAN* Management Adapter on my box.
This
Does anyone know how (if possible) to do three way calling on the
UIP-200? There doesn't seem to be much info about this phone, but all
the feature lists I've read says it can do conference calls. I can't
seem to do it, though. Any help would be appreciated.
--
Seth et lux in tenebris lucet
Ah... Looks like I have the 500's...Sorry.
Ty Purcell
-Original Message-
From: Derek Listmail Acct [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 09, 2004 12:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycon IP 300 SIP vs
In July I bought one from CDW for $280.75.
Ty
-Original Message-
From: hank smith [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 09, 2004 2:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940
what is the price
A local vendor here carries IP500s for sub $200. Right now they are out
of stock, but he has more coming in. If you want his contact info msg me
off list.
-Tim
-Original Message-
From: Ty Purcell [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 09, 2004 2:45 PM
To: Asterisk Users
Hank,
IP500's are 199$ USD at voipsupply.com
and
Cisco 7940 are 295.99$ USD at voipsupply.com
Thanks,
Matt
hank smith wrote:
what is the price range in us dollars?
- Original Message - From: Jody N. Rudolph [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Is anyone familiar with the Trollphone's LCR package?
There is a field in the egress table labeled substitue. Placing a $1 there
results in the correct dial extension being passed. However how is this
field used to substitute replacement dial extensions... in other words as
an example, lets
How do you get Cepstral working, they only offer windows versions. do I
have to complie it to linux?
http://www.cepstral.com
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Greetings,
I've just installed Asterisk RC2 on a new RedHat 9 box (fully patched).
The build and install went fine, but after starting Asterisk, I get the
following messages in /var/log/asterisk/messages. Restarting Asterisk
produces the same errors:
-
Sep
It sounds like Skinny couldn't load- doesn't sound fatal.
On Thursday 09 September 2004 08:01 pm, [EMAIL PROTECTED] wrote:
Greetings,
I've just installed Asterisk RC2 on a new RedHat 9 box (fully patched).
The build and install went fine, but after starting Asterisk, I get the
following
On Thu, 09 Sep 2004 13:59:20 -0600, TELUX [EMAIL PROTECTED] wrote:
How do you get Cepstral working, they only offer windows versions. do I
have to complie it to linux?
http://www.cepstral.com
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Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi,
I posted to this
list couple of days ago, that my astcc is not writing the card balance to the
mysql database.
http://lists.digium.com/pipermail/asterisk-users/2004-September/061645.html
I just want to ask
this question one more time before creating a bug note in "mantis". Since the
Hi All,
I am setting up two boxes with asterisk. Box A has a T100P.
It is working I can put calls in the outgoing spool directory
and call extensions or outside numbers and do a playback of
demo-congrats (not sexy but good for an example).
Now I have a second Box b. Box B has no hardware just
Seth Mattinen wrote:
Does anyone know how (if possible) to do three way calling on the
UIP-200?
The UIP-200 currently doesn't support this, which is a shame.
I typically create a meetme room for every sip extension (ie: 8XXX where
XXX is the exten number). Users can then transfer callers to
Works for me, follow the instructions closer. :)
Storm D. J. Petersen wrote:
Hi,
I cannot seem to accept incoming calls from FWD using IAX2. I followed the
directions posted at www.fwd.pulver.com/advanced/iax I can make outgoing
calls fine using IAX via FWD. When someone calls me from FWD I
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: September 9, 2004 1:07 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] IAX2 dropping call?
Hello all,
I updated from CVS 3 days ago and now my IAX2 gateway is dropping
calls without warning.
On Thu, 2004-09-09 at 14:59, TELUX wrote:
How do you get Cepstral working, they only offer windows versions. do I
have to complie it to linux?
http://www.cepstral.com
They have a linux version for purchase on their web site.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
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