Karl J. Vesterling wrote
(B
(B Get an HTML capable E-Mail Program.
(B They've been freely available for nearly
(B 10 years now and on just about every platform
(B that's got more than a 16 bit bus.
(B
(BJust because some indecent folks have decided to put
(Bindecent features into mail
Hi,
On Sun, Aug 01, 2004 at 09:13:52PM +0200, Massimo De Nadal wrote:
...
[from-ISDN1]
exten=s,1,Wait(1)
exten=s,2,Dial(Sip/cisco1Sip/xlite1,30,tTr)
exten=s,3,HangUp
The problem is that when I receive a call, I can't see the CallerID neither
on the Cisco 7940 nor on the X-Lite client.
This sounds great in my opinion, I am looking forward to hearing more
about it, as far as list etiquette it may be off topic but many of us
do use the ciscos and it could be a great feature we can offer our
clients.
mitchel
On Thu, 29 Jul 2004 09:28:09 +0200, Holger Schurig
[EMAIL PROTECTED]
- STUN support (SIP)
No clue, I was testing the phone only in my local network.
- Three-way-calling (SIP MGCP)?
a) I did not get the phone running in MGCP mode.
b) the phone doesn't have a TRANSFER button, so I assumed that it
doesn't have any built-in transfer capability
- did you get
- quality of the speakerphone better than the horribe GS speakerphone?
The LOUDSPEAKER is quite ok.
But the phone doesn't seem to have a microphone built into the chassis. So
when you press the loudspeaker button, people in the office can listen to
the call, but they can't participate and
Hello,
I read the previous postings in asterisk-users mailinglists and I didnt
found any postings related to my problematic topic.
Problem:
If I need to set different pridialplan for different channels. For example:
group1 has first 15 channels and all calls what are sendt via this group
are
On Sun, 1 Aug 2004 14:50:26 -0400 (EDT), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
On Sun, 1 Aug 2004, Trevor Peirce wrote:
Hello,
I know not too long ago I saw /something/ _somewhere_ about an
adjustment to call parking that allowed blind transfers from SIP phones
to park a call and
Hello Holger,
Friday, July 30, 2004, 11:40:25 AM, you wrote:
HS At the end of this website is the AT-323 that I have here for test. I
Where did you purchase this phone? I tried to search Google but
didn't find any shops selling AT-323.
--
Best regards,
Oleg
On Mon, 2 Aug 2004, Jason Williams wrote:
On Sun, 1 Aug 2004 14:50:26 -0400 (EDT), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I use Brian's Valet Parking on our system.
exten = 700,1,ValetParkCall(7${CALLERIDNUM}|mylot|60|${CALLERIDNUM}|1|phones)
exten =
On Mon, 2 Aug 2004 04:50:08 -0400 (EDT), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
On Mon, 2 Aug 2004, Jason Williams wrote:
On Sun, 1 Aug 2004 14:50:26 -0400 (EDT), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I use Brian's Valet Parking on our system.
exten =
I've got a problem connecting to Cisco call manager.
I dial a numder and hear only ringing
h.323 debug shows this
Allowed Codecs:
Table:
G.711-ALaw-64k{sw} 1
Set:
0:
0:
G.711-ALaw-64k{sw} 1
-- Making call to [EMAIL PROTECTED]
== New H.323 Connection created.
-- 6129 is
hi!
I'd like to send a FAX/SMS message using asterisk. But I should be able to
detect the end terminal before sending. So when the destination
terminal is a FAX terminal the asterisk should call,
txfax
or if it's not it should call
SMS
achievable ??
bit123.
hi list,
i want to convert all none SIP calls to h323 and send them to our GnuGK
Gatekeeper.
with my setup (attached) i called the number 5678 and got the following
error msg:
Error msg:
Jul 29 10:19:45 WARNING[114696]: chan_sip.c:457 __sip_xmit: sip_xmit of
0x81210dc (len 635) to 0.0.22.46
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 02 August 2004 04:58 am, Jason Williams wrote:
On Mon, 2 Aug 2004 04:50:08 -0400 (EDT), [EMAIL PROTECTED]
And now it magically works with three digit extensions. Do you need me to
paste the config for four digit extensions as well?
Hello,
I've been trying to get the CDR to work but when I do make it prints:
cdr_addon_mysql.c:33:19: mysql.h: No such file or directory
cdr_addon_mysql.c:33:19: errmsg.h: No such file or directory
and then stops.
Is this something to do with MySQL or with asterisk? I have the latest
version of
Any ideas where I can get these files from?
You need to install the development files for MySQL, not just the app.
Depending on your distro, they could be named mysql-dev, libmysql-dev or
similar.
___
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[EMAIL PROTECTED]
Tom Lawrence [EMAIL PROTECTED] wrote:
_/_/_/_/_/ _/_/_/_/ _/_/_/_/_/
_/ _/_/_/_/ _/ _/ _/
_/ _/_/_/_/ _/ _/ _/
_/ _/_/_/_/ _/ _/ _/
Resistance is futile. :-)
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/
Thorsten Huber wrote:
Hi,
On Sun, Aug 01, 2004 at 09:13:52PM +0200, Massimo De Nadal wrote:
...
[from-ISDN1]
exten=s,1,Wait(1)
exten=s,2,Dial(Sip/cisco1Sip/xlite1,30,tTr)
exten=s,3,HangUp
The problem is that when I receive a call, I can't see the CallerID
neither
on the Cisco
Dear List,
I have made an update to asterisk RC1 - all works well :)
but I am getting all the time an error message:
Aug 2 13:34:57 NOTICE[15375]: sched.c:221 sched_settime: Request to
schedule in the past?!?!
Aug 2 13:34:57 NOTICE[18446]: sched.c:221 sched_settime: Request to
schedule in
Hi,
On Mon, Aug 02, 2004 at 01:20:32PM +0200, Massimo De Nadal wrote:
we had similar problems and fixed them by setting the CIDName to the
CallerID:
[from-ISDN1]
exten=s,1,Wait(1)
exten=s,2,SetCIDName(${CALLERID})
exten=s,3,Dial(Sip/cisco1Sip/xlite1,30,tTr)
exten=s,4,HangUp
Thank
Hi,
On Mon, Aug 02, 2004 at 01:20:32PM +0200, Massimo De Nadal wrote:
we had similar problems and fixed them by setting the CIDName to the
CallerID:
[from-ISDN1]
exten=s,1,Wait(1)
exten=s,2,SetCIDName(${CALLERID})
exten=s,3,Dial(Sip/cisco1Sip/xlite1,30,tTr)
Hi folks,
Anybody making fax-on-demand with * ?
Isamar
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Hi Trevor,
Trevor Peirce wrote:
Hello,
I know not too long ago I saw /something/ _somewhere_ about an
adjustment to call parking that allowed blind transfers from SIP phones
to park a call and still be able to hear the parking lot stall number.
Unfortunately, I have no idea where I saw that
All,
What i'm trying to do is setup a windows DUN connection via my asterisk box
and over PSTN or VOIP to my work. What I hoped i'd find was a vitual modem
driver for windows 2000 that wouldtalk over sip to my asterisk box and then
act like a normal modem so I can dial out from that to our RAS
I don't know of a program for doing this, but then again I have never looked for
one :-). I wonder if a sipura or an iaxy would do the job though.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Jon Creasey [EMAIL PROTECTED]:
Hi
I'd rather avoid cutting the case.too much work involved and too much risk of an
error.
P
-Original Message-
From: Kevin Walsh [mailto:[EMAIL PROTECTED]
Sent: Sunday, August 01, 2004, 3:31 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Cisco 7960 backlight update
Hi,
On Mon, Aug 02, 2004 at 01:20:32PM +0200, Massimo De Nadal wrote:
we had similar problems and fixed them by setting the CIDName to the
CallerID:
[from-ISDN1]
exten=s,1,Wait(1)
exten=s,2,SetCIDName(${CALLERID})
exten=s,3,Dial(Sip/cisco1Sip/xlite1,30,tTr)
Hello,
I am planning to use my Quickent PhoneJack card with Asterisk. It appears to
be the only way to use a Quicknet Card with SIP. Following some of the
postings it should be possible. However I cannot get it to run.
I have installed the card successfully running Linux 2.6.6 with the native
Yes I've implemented a simple web interface that generates a . call file
that faxes generated .tiff files a Crontab checks against a
database to generate the tiffs and .call files.
B
Isamar Maia wrote:
Hi folks,
Anybody making fax-on-demand with * ?
Isamar
Does anyone know of
any provider(s) that can provide DID's for the Czech
Republic?
Regards,
-Steve
I have no doubt that this would work but I don't like the idea of all those
conversions rather than just IP most of the way.
Jon
- Original Message -
From: Jonathan Moore [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, August 02, 2004 1:04 PM
Subject: Re: [Asterisk-Users] Win2000
May be that information can help...
On definity
display dialplan analysis Page 1 of
3
DIAL PLAN ANALYSIS TABLE
Percent Full:
6
Dialed Total Call Dialed
On Sun, Aug 01, 2004 at 04:03:58PM -0500, Martin Keding wrote:
I just brought in a 480i for testing. It is VERY bare basics! Part of the
Web interface still doesn't work (doesn't show you the sip setting) and
barely has any other settings available. Also does not have any features for
NAT. The
Probably best to install MySQL 4 (free) available from
http://dev.mysql.com/downloads/mysql/4.0.html
Under Red Hat or Fedora, for example, I find that these libraries are
necessary:
If you use rpm's, install by using rpm -U for each, in this order:
* shared-compat (may be called something
Hi,
I want to know, if someone has tried to use clustering
in asterisk to increase its scalability ??
If yes, how easy it is to cluster and whats the
procedure?
Varun
__
Do you Yahoo!?
Yahoo! Mail is new and improved - Check it out!
After also changing
extensions.conf
exten = 4321,1,Dial,Phone/phone0
and completely stopping and restarting asterisk (reload was not enough ?) I
can call my phone with dial 4321 and it rings, it also provides a dialtone.
Fritz
-- Weitergeleitete Nachricht --
Subject: Help
Hi,
I Have a problem here, if anyone know a method to avoid please tell me .
Using * with the option canreinvite=yes i can in theory tell to my * box, send RTP Packet directly from one
Sip device to another one, then "In Theory", i will not use my own internet connection.
So this mean that
I wanna do it through IVR.
I know how to create the .call files for normal outbound calls but how to
attach the .tiff files ?
Isamar
On Mon, 2 Aug 2004, Brian McManus wrote:
Yes I've implemented a simple web interface that generates a . call file
that faxes generated .tiff files a
Charles,
I hate to say it, but the server that runs our web site is located in
Sollentuna, Sweden which is GMT. The browser and it's Javascript calculator
were using your local clock. Sorry if that threw you off.
Thanks,
Steve
Steven Sokol
Owner/Manager
Sokol Associates, LLC
Phone:
I think the reason is because the telephony equipment of your telco is
still analog.
(In belgium it was the same, until they started replacing all the old stuff
with fancy digital things.).
At 17:30 29/08/2004, you wrote:
Hi,
in Spain that process is correct. If you setup a communication
On Mon, 2 Aug 2004, Areski wrote:
I have made an update to asterisk RC1 - all works well :)
but I am getting all the time an error message:
Aug 2 13:34:57 NOTICE[15375]: sched.c:221 sched_settime: Request to
schedule in the past?!?!
Aug 2 13:34:57 NOTICE[18446]: sched.c:221
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
Private network serving the local user (1)
Ext: 1 Cause: Incompatible destination (88), class =
Invalid message (5) ]
Here's how I've got mine set up, maybe it will help, it's a little
different then
Hi Everyone,
I'm new to asterisk and trying to get together the hardware to run a few POTS
phone extentions and one or two POTS lines for starters. For these low port
counts, I could just go with FXS and FXO cards, but...
I can get a Cisco MC3810 with a mixture of FXO and FXS ports, the MC3810
...now, we don't wish them ill, but Vonage seems to have been out of
commission for quite a little while. Website is excruciatingly slow,
log-in fails, hard-line and soft-line are out (alert-tone or subscriber
not in service) -- even Network failover is failing. Outgoing calls
return fast-busy.
I'm curious as to folks experiences in selling asterisk-based solutions.
In sales-speak, what are the common compelling reasons to buy?
I can think of the following potential ones, but I'm keen to find out what
seems to work in practise:
- Customer wants to cut cost of calls, implements *
Hello again!
Just wondering if any one else has had a problem with stop and starting
asterisk?!? If I do it say 5/6times without restarting the computer then it
crashes. This doesn't seem normal to me, could this be because I'm running
fedora core 2? I know there's problems with using fedora to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Monday 02 August 2004 11:09 am, Wayde Nie wrote:
Hi Everyone,
I'm new to asterisk and trying to get together the hardware to run a few
POTS phone extentions and one or two POTS lines for starters. For these low
port counts, I could just go
Hi -
I'm new to the whole Asterisk/IP phone phenomenon. The documentation
on Asterisk is great, but the documentation on the handsets seems to be
somewhat sporadic. My questions on handsets:
1. Which handsets support multiple simultaneous calls? I know that
the Cisco 7960 supports 6, the
On Mon, 2 Aug 2004, David Gurr wrote:
I'm curious as to folks experiences in selling asterisk-based solutions.
In sales-speak, what are the common compelling reasons to buy?
Those are good reasons, but one compelling reason is that it's pretty
inexpensive to set up a system. We sell Avaya
I'm having problems in dialing numbers over SIP that include characters from
the UK international phone number conventions.
I have my contacts in Outlook, with the numbers represented as:
+countrycode (area code) numberpart numberpart
eg:
+44 (20) 7834 1234
or:
+1 (801)
Hi,
A potential customer would like to be able to do this: If a call comes in
for an employee who is on the phone, allow the front-desk to push the caller
in a queue directly to the employee. Now, this is easily done by using
queues, but I am curious: What is the performance impact on a system
On Mon, 2 Aug 2004, Carlos Arnt wrote:
I Have a problem here, if anyone know a method to avoid please tell me.
Using * with the option canreinvite=yes i can in theory tell to my *
box, send RTP Packet directly from one Sip device to another one, then
In Theory, i will not use my own internet
Hi
Can anyone give suggestion why we need STUN while using asterisk behind the NAT.
Regards
Shan.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi there,
i'm in debian sid 3.1 with kernel 2.6.7, * last cvs chan_capi 0.3.4b; nt1+ with 2
bri in ptmp (http://www.voip-info.org/tiki-index.php?page=DDI)
i tried to install avm c4 following step by step
-Original Message-
From: Florian Overkamp [mailto:[EMAIL PROTECTED]
Sent: Monday, August 02, 2004 11:41 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Performance of queues
Hi,
A potential customer would like to be able to do this: If a
call comes in for an employee
Message: 9
(BJay Milk wrote:
(B
(B Let's see how they deal with that in their
(B oh-so-controlled environment
(B ;-)
(B
(BYou naughty boy!
(B
(B;-)
(B
(Brgds
(Bbenjk
(B
(B
(B--
(BSunrise Telephone Systems Ltd
(B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan
(B
Hi there,
I am using bri-stuff.0.1.0-RC2k and it seems that things didn't become
better. I have got lots of dropouts on the IAX2 link (no matter if
jitter buffers are enabled).
Further the MP3Player application does not playback streams like
http://somestreamserver/somestream. It stops saying:
Can someone tell me where I can get just app.c from. Mine somehow got
corrupted, and no updates or anything else will fix it. I just need the one
file from the latest cvs. 8-1-04. Please help
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[EMAIL PROTECTED]
Delete it and cvs update will retrieve it.
-Original Message-
From: AJ Grinnell [mailto:[EMAIL PROTECTED]
Sent: 02 August 2004 17:33
To: Asterisk
Subject: [Asterisk-Users] App.c
Can someone tell me where I can get just app.c from. Mine somehow got
corrupted, and no updates or anything
Maurizio Marini wrote:
[controller1]
msn=0xx
...
when i issue an outside call i get:
-- Executing Dial(SIP/sip1-07f4, CAPI/0721xx:bBYEXTENSION:1) in new stack
-- data = 0721xx:b0721950396:1
-- capi request omsn = 0721xx
Aug 2 17:53:02 NOTICE[1224547248]:
Thanks, Ill give that a try. My * box is going crazy right now. Anyone know
if yesterdays updates would cause ALL of my Sipura SPA2000s to loose
registration randomly and not come back up?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Hanselman
On Mon, 2004-08-02 at 12:34, Steve Hanselman wrote:
Delete it and cvs update will retrieve it.
cvs update -C app.c would also work. -C retrieves the clean copy
from the repository and saves your local changes into another file.
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Is anyone else having problems with Sipuras not being able to re-register to
Asterisk after applying the cvs update last night? Just curious if I need to
roll back or take all of my Sipuras out back and beat them.
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STUN (RFC-3489) is an UNSAF type network protocol (see RFC 3424) that is
used to discover UDP address and port
bindings across network address translators.
(a) Currently Asterisk only supports static configuration of the
external IP address of a NAT.
You need to discover it manually by other
Someone giving DID for Spain?
Thanks in advance
Adrià Vidal
mailto:adriavidal at telefonica.net
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Hi Karl,
I'm suffering with the problem you outlined in (a) regardless of a STUN Server
being used.
Is their anyway around this?
Cheers,
Sahil
Quoting Karl Brose [EMAIL PROTECTED]:
STUN (RFC-3489) is an UNSAF type network protocol (see RFC 3424) that is
used to discover UDP address and port
Sunrise Ltd wrote:
Ming-Wei Shih wrote:
I have a login on the wiki but IMHO this
does not belong to the wiki, it
should be in the src.
It belongs on the Wiki for as long as it takes to get it
into the CVS, because that's where people will be looking
for help.
Once the modifications
Hi Eric,
On Sat, 2004-07-31 at 17:55, Eric Bart wrote:
I don't understand why sipura can do consultative transfer
and why grandstream can't. They're both SIP, aren't they ?
They use different sip stacks... and yes, they are both sip.
Maybe the sipura transfer is using a sip
Hi,
what are the Systemrequirements for Asterisk with
SIP?
Moritz Beierlein
AJ Grinnell [EMAIL PROTECTED] wrote:
Is anyone else having problems with Sipuras not being able to re-register
to Asterisk after applying the cvs update last night? Just curious if I
need to roll back or take all of my Sipuras out back and beat them.
My SPA-2000 is fine (today's CVS).
Hi,
what are the Systemrequirements for Asterisk with SIP?
Moritz Beierlein
Hi Mortiz,
The system requirements are not really a matter of Asterisk with Sip.
Posting some more information in regads to number of Sip clients, codec
requirements, number transcoding streams, etc would be more
I will do it and let you know.
On Sat, 31 Jul 2004 17:36:22 -0500 (CDT), Bartosz Wegrzyn [EMAIL PROTECTED]
wrote:
I am ready to close that topic.
Finally, I replaced my router from Multitech for Linksys.
It solved all the problems related to NAT and incoming calls issues.
My router model
Ok, thanks
- Original Message -
From: Michael Manousos [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, August 01, 2004 10:37 AM
Subject: Re: [Asterisk-Users] asterisk-oh323-0.6.3a
M. Willigs wrote:
Hi there.
I thy to compile asterisk-oh323-0.6.3a but it fail in the make
Did anybody experience and problems with broadvoice today?
I could not call into my box till 1.30 pm.
I was always redirected to my mailbox at broadvoice.
After 1.30 everything started to work again.
Bart,
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[EMAIL
It works with ZAP FXO, Sipuras and Grandstream phones. The sipura is
able of 3way conferences by itself. The consultative transfer is a kind
of 3way conference for the Sipura..
So it seems that the others parties keep running through the sipura,
even in a consultative transfer. So you can't
My probles started again.
2.30.I cannot register.
Connected to Asterisk CVS-HEAD-07/27/04-12:25:03 currently running on
lexon (pid = 12654)
-- Remote UNIX connection
Aug 2 14:25:20 WARNING[-1147659344]: chan_sip.c:673 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED] for
seqno
On Monday 02 August 2004 15:28, Bartosz Wegrzyn wrote:
My probles started again.
2.30.I cannot register.
Surely there are other providers to investigate, or a customer service desk at
Broadvoice to complain to.
-A.
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Asterisk-Users mailing list
2.34 pm they are back up and running.
What is going on???
My probles started again.
2.30.I cannot register.
Connected to Asterisk CVS-HEAD-07/27/04-12:25:03 currently running on
lexon (pid = 12654)
-- Remote UNIX connection
Aug 2 14:25:20 WARNING[-1147659344]: chan_sip.c:673
Hi Eric,
On Mon, 2004-08-02 at 16:26, Eric Bart wrote:
It works with ZAP FXO, Sipuras and Grandstream phones. The sipura is
able of 3way conferences by itself. The consultative transfer is a kind
of 3way conference for the Sipura..
So it seems that the others parties keep running
Absolutely, there is VoicePulse, BroadVox, Nufone, etc. Also If you
want exceptional stability and don't mind paying the man ATT also has
business and residential VoIP service (it's a bit spendy but very
reliable, and for business a rep told me they have 100% Service Level
agreements, if they
Anyone had experience 'marrying' the two?
We had setup * to front end Artisoft's Televantage.
It works with some issues need to be resolved:
- Inbound calls could not properly handled and routed by Televantage's
Call Classifier. It goes directly to the Televantage's default auto
attendant.
As far as I know all those comanies want to be reliable.
You talk like broadvoice is one of the comanies that goes down very often.
And as a alternative you suggest to switch.
I don't think that this is the way I or other people should go.
Broadvoice needs to be 100% up and I now they want that.
Hi
(B
(Banybody who would like to test drive a pre-release of the
(Bfirst OSX Assistant please visit the Wiki ...
(B
(Bhttp://www.voip-info.org/tiki-index.php?page=Asterisk+MacOSX+Support
(B
(Bif you find any bugs please let me know by email: benjamin
(B(at) sunrise-tel (dot) com.
(B
[EMAIL PROTECTED] wrote:
Hi Karl,
I'm suffering with the problem you outlined in (a) regardless of a STUN Server
being used.
Is their anyway around this?
It's not a fault of the STUN server.
Yes, with a little patience there will be a way around this. We are
close to releasing STUN support
Sipura is limited to 3way conferences (or 2 line appearences)
so you cannot have two 3way conf (or consultative transfer)
at the same time.
If you would like to have many calls onhold/waiting, you can use
asterisk with parking or valet, or even call queues. If you can afford
the hardware,
For the record, That wasn't the intent of my post, he did ask if there
were alternatives, and I mentioned there are alternatives.
I agree they all would like to be up 100%, and since I've never used
BroadVoice, I can't talk negatively or positively represent them, as I
have no experience using
I'm trying to get call parking working with the lighted buttons on the
SNOM 200. I have set the 5 buttons to Park Orbit, for extensions 700-704.
Pressing the first button (x700) does park the call. However, the
remaining buttons (x701-704) don't allow me to pick up parked calls, or
show
On Monday 02 August 2004 16:20, Brian McManus wrote:
For the record, That wasn't the intent of my post, he did ask if there
were alternatives, and I mentioned there are alternatives.
It was the intent of my post; There have been what, 50 messages about
Broadvoice being down on this list? It's
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, August 02, 2004 1:32 PM
Subject: Re: [Asterisk-Users] Today's possible problems with Broadvoice
On Monday 02 August 2004 16:20, Brian McManus wrote:
For the record, That wasn't the
Hi everybody.
I install the new Asterisk 1-RC1 on my machine and I can't make the Playback
function works through chan_h323 whith version 7.x's configs files
Any ideas?
Thanks in advance.
M. Willigs
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[EMAIL PROTECTED]
This is my first post, so please feel free to direct me to
another list if needed.
I am in the early stages of researching Asterisk. I
administer a small Avaya Definity G3 switch (~400 users).
Can anyone point my to resources/documents/actual
implementation notes of using Asterisks
Hi,
-Original Message-
This is my first post, so please feel free to direct me to
another list if needed.
Nope, this list is appropriate. There is some startup reading to do, however
:-)
I am in the early stages of researching Asterisk. I
administer a small Avaya Definity G3
Yes, check the Wiki, I believe there are some Avaya notes in there
that speak of the Definity. http://www.voip-info.org. Do a search for
Avaya, or definity and that should direct you to at least some
information.
As far as integration with Voicemail, that is something I am currently
battling with
Hey Brian, my name is Brian too. I too administer a Definity.
I use Asterisk for personal use, but to use Comedian Mail with
Asterisk should be about the same as using anything else like Intuity
Audix.
Create your hunt group in the Definity, and then assign how ever many
analog ports you want,
Hello,
we had a running configruation where asterisk passed
the phone number and the ddi to the pstn (i.e. 595-431)
Now only the rootnumber arrives:
5950
I do not know, what to do. I tried to use callingpres
(now i am just hiding every number, because 595-0 is no valid extension..) but
On Monday 02 August 2004 16:59, Chris Shaw wrote:
Grant you, e-mails that are sent to the list to bitch and moan about BV
being down and I'm gonna take my business somewhere else... those don't
belong in this list, those don't even belong in -biz... I'd be surprised if
ANYONE would read
On Mon, 2004-08-02 at 16:23, ePyron Felix Deierlein wrote:
With kind regards
Kindly do not send email in HTML. Your message was particularly
offending for having changed the font color to a green that was harsh on
the eyes and reducing font size by about 2 points.
Please understand that just
Hi everyone
I'm very, very, very, ... very new to Asterisk and I
need some help with the ParkAndAnnounce command.
Here's what I would like to do. I would like to
specify an extension in the extension.conf file which
is using the ParkAndAnnouce command (something like
this)
exten =
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, August 02, 2004 2:48 PM
Subject: Re: [Asterisk-Users] [RANT] Today's possible problems with
Broadvoice
On Monday 02 August 2004 16:59, Chris Shaw wrote:
Grant you, e-mails that are
On Monday 02 August 2004 16:20, Brian McManus wrote:
For the record, That wasn't the intent of my post, he did ask if there
were alternatives, and I mentioned there are alternatives.
It was the intent of my post; There have been what, 50 messages about
Broadvoice being down on this
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