Use RC16. This seems to solve our issues on a UK ISDN2e line.
Rgds
Tim
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julien Levi
Sent: 02 April 2004 01:17
To: [EMAIL PROTECTED]
Cc: Jens-Uwe Junghanns
Subject: [Asterisk-Users] quadBRI card installation
For anyone who is interested..
I downgraded my system to..
asterisk-0.7.2
libpri-0.5.2
zaptel-0.8.1
+asterisk-addons
..and its all working again...
Later..
WipeOut wrote:
Hi,
I have just built my home Asterisk box into a better PC that became
available (still only a P2 350 but it only has
Title: Message
Well,
the best new product that I saw was the IAX (or was that SIP?) WiFi 2500 hard
phone thatMark demo'd for everyone at the Mexican joint after
dinner...Hans, do youhave a pic of that sleek, modern yet dare I say haute
couture look? I think that Digium could take over the
Hi,
how can I checkout ztdummy?
Thank for you help.
Felix Deierlein
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G'day Raymond,
On Thu, 1 Apr 2004, Raymond McKay wrote:
I have seen a few postings in the past regarding the interop of Asterisk and
the Cisco 7920 WiFi phone. To date, I have not seen a definitive method to
getting the phone working. Assuming someone has this actually working, can
that
Felix,
how can I checkout ztdummy?
Thank for you help.
Checkout of cvs the zaptel source then follow these instructions:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy
JR
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Hi,
thanks.
how can I checkout ztdummy?
Thank for you help.
Checkout of cvs the zaptel source then follow these instructions:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy
I have tried to follow, but I did not know, wich modul I had to check out..
Bye
Felix
I downloaded the app and for the most part have it going.
I have not yet managed to get it to accept the password in the flash
widget that appears as if it wants to accept it.
I wonder about browser-related problems in that respect: I'm running
fairly recent Mozilla.
I have also hacked the
On Fri, 2004-04-02 at 10:43, ePyron Felix Deierlein wrote:
I have tried to follow, but I did not know, wich modul I had to check out..
Checkout the Zaptel CVS module. Edit the Makefile in the Zaptel dir to
uncomment the ztdummy source.
--
Glen Gray [EMAIL PROTECTED]
On Fri, 2004-04-02 at 10:43, ePyron Felix Deierlein wrote:
I have tried to follow, but I did not know, wich modul I had to check
out..
Checkout the Zaptel CVS module. Edit the Makefile in the Zaptel dir to
uncomment the ztdummy source.
--
Glen Gray [EMAIL PROTECTED]
Can someone help me what went wrong with this call?
This call was initiated from dev/ttyI0 device on my asterisk server to
mediatrix unit. Mediatrix unit user received the call and call started.
I can hear them OK but they can not hear me correctly (cut-off sound,
noise). Call was finally hunged
Simon Brown wrote:
I have quite successfully set up the Services button to work on the 7940
running SIP.
I have a metric-imperial converter, a foreign exchange rate calculator, a
calendar etc available to users.
The XML is really fussy though.
Could you share these example applications?
Thanks,
Alternatively, put it somewhere where we can all get at it :D
Andy
*** REPLY SEPARATOR ***
On 02/04/2004 at 06:52 Raymond McKay wrote:
I am using one version of their chan_sccp with a 7960, and can vouch for
its functionality there. If you strike out finding an
Vic Cross ([EMAIL PROTECTED]) wrote:
G'day Raymond,
On Thu, 1 Apr 2004, Raymond McKay wrote:
I have seen a few postings in the past regarding the interop of Asterisk and
the Cisco 7920 WiFi phone. To date, I have not seen a definitive method to
getting the phone working. Assuming
G'day Jeremy,
On Fri, 2 Apr 2004, Jeremy Bogan wrote:
Line 1 is the home line, I want to give my DECT cordless phone system
it's own extension and this phone will ring when this line is called.
I'd like outgoing calls made from the DECT system only to be made from
that first line.
Easy.
Best Web Hosting Resource
Hi All,Best Web Hosting ResourceFor musiconhold.conf file, how can I play the track01.mp3 as default musicwhen it is on hold?Best Web Hosting ResourceBelow conf is it correct?[classes]default = mp3:/var/lib/asterisk/mohmp3/track01.mp3Best Web Hosting ResourcePlease
Office Space Quotes : Get Office Space Quotes
Hi All,Office Space Quotes : Get Office Space QuotesFor musiconhold.conf file, how can I play the track01.mp3 as default musicwhen it is on hold?Office Space Quotes : Get Office Space QuotesBelow conf is it correct?Office Space Quotes : Get Office
Il 22:52, giovedì 01 aprile 2004, Nicolas Gudino ha scritto:
http://sip.house.com.ar/operator
Best regards,
I've seen that keynames are very strictly.
The problem is that for example CAPI channel, change the name every time with
a serial number
canal: SIP/GS1
canal: MGCP/[EMAIL PROTECTED]
Hi Jeremy,
Jeremy Hall wrote:
Actually, the short answer any more is yes, you can use a modem.
Cool! that could make my life easier when setting up a demo system to
sell Asterisk to my bosses... :-)
I know it is better for several reasons to use an actual Digium X100P.
The main reason being
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-Original Message-
http://sip.house.com.ar/operator
I love these types of applications that show off the capabilities of *.
This was easy to get up and running for my SIP channels, but for some
reason my PRI (ZAP/1 through ZAP/6) aren't showing up. Has anyone else got
this working for
if you don't give them the pass code they can't hang-up or transfer calls
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adams, Gavin
Sent: Friday, April 02, 2004 7:30 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
this is not where to send your unsubscribe to
!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Altus Snyman
Sent: Friday, April 02, 2004 7:20 AM
To: asterisk
Subject: [Asterisk-Users] UNSUBSCRIBE
just type it in it will remain until you restart your browser. ( it does
not disappear and you do not have to hit enter or anything like that)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian Capouch
Sent: Friday, April 02, 2004 3:45 AM
To: [EMAIL
My apologies to the list members, I sent the mail by mistake to all of you,
while my intention was to send it to Matt Ridell only.
I also made a typo in the naming convention for IAX2, you have to remove the
slash after IAX2.
If you have problems/questions/bug reports with the operator panel,
Title: Message
Is
that wifi phone available? If yes how much and when? I am looking to purchase a
large quantity of wifi phones. I have a few questions on making calls with these
phones and how the accounting of the calls would go. Thanks.
Sincerely,Stephen KarringtonDreamtime.net
Being able to have more buttons as well as changing the button size
would be useful.
On Fri, 2004-04-02 at 08:04, Nicolas Gudino wrote:
My apologies to the list members, I sent the mail by mistake to all of you,
while my intention was to send it to Matt Ridell only.
I also made a typo in the
It appears Asterisk can handle DTMF inband on only a limited selection of
formats, of which G.729 is not one. The issue appears to be something
involving short data -- whatever that is. (I'm inferring all this from
looking at dsp.c in the vicinity of the error message I was getting, which
pointed
Hi Eric,
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
Sent: Friday, April 02, 2004 11:17 AM
Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
Being able to have more buttons as well as changing the button size
would be useful.
What screen resolutions do you
Does anybody know of any software that can show the status of voicemail
messages? Or at least provide a visual indication that I have new voicemail?
Right now I am using Gnophone and I'm checking manually.
Thanks in advance.
--
Christopher Lewis
We run at 1600x1200, 96 buttons would be useful.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nicolas Gudino
Sent: Friday, April 02, 2004 9:26 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
I put an externip=xxx.xxx.xxx.xxx in my sip.conf so I can register with FWD
from behind a NAT
With this entry my PSTN calls have a problem in that the other party cannot
hear me - I can hear them.
It does not matter whether I make the call or the other party does.
Any ideas ?
TIA
Simon
we also would require more buttons, at least 40, can we get a multipage
view. right know I run multiple servers on the same page to get the effect
of having 3 pages.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nicolas
Gudino
Sent: Friday, April 02, 2004
Bryce Nesbitt (mailing list account) wrote:
I received a recommendation to check out Asterisk, as a platform to host
a simple DTMF response system, something like:
Setup up VoIP endpoint on Linux/FreeBSD system
Answer incoming VoIP phone calls
User enters 100#, perl script plays back foo
Hi all,
I installed all needed RPMs by GnoPhone to be installed without problems
but when attempting to install GnoPhone itself I get this message:
# rpm -Uvh gnophone-0.2.4-1.i386.rpm
error: Failed dependencies:
mozilla = 0.9.2 is needed by gnophone-0.2.4-1
libgtkembedmoz.so is
Mireia Munoz de jesus [EMAIL PROTECTED] wrote:
My gateway accepts G.711, but not my Grandstream 100 series SIP phone
Mine does. It is termed PCMU and PCMA in the Grandstream setup.
Doug
--
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
DaveTiptonInfrastructureArchitect817-858-9841VoiceEuless,TXHamRadioCallSign:W3DMT--Thedefinitionofinsanityisdoingthesamethingoverandoverandexpectingdifferentresults.--BenjaminFranklin-Original
It's not asterisk, its the codecs. Codecs other than ulaw and alaw will
distort continuous tones like DTMF.
On Fri, 2004-04-02 at 08:22, Jim Rosenberg wrote:
It appears Asterisk can handle DTMF inband on only a limited selection of
formats, of which G.729 is not one. The issue appears to be
Title: Message
I am just an
Asterisk newbie doing a test install. I am using 2 X-Lite clients and
haveconfigured them according to the wiki on voip-info. A warning is
still displayed on the Asterisk server console saying that I should disable
RFC3389 on the client, even after I changed the
On Friday 02 April 2004 16:01, Martin Mielke wrote:
Hi all,
I installed all needed RPMs by GnoPhone to be installed without problems
but when attempting to install GnoPhone itself I get this message:
# rpm -Uvh gnophone-0.2.4-1.i386.rpm
error: Failed dependencies:
mozilla = 0.9.2
My users usually use 800x600 and I would need as many buttons as can fit
on that screen. 8-) One of my servers currently has 18 Zap channels and
6 IAX2 peers. I switched my laptop to 600x600 and the bottom row of
buttons is cut partially off.
Another feature, which would be nice is if you
-Original Message-
Hi Jeremy,
Jeremy Hall wrote:
Actually, the short answer any more is yes, you can use a modem.
Cool! that could make my life easier when setting up a demo system to
sell Asterisk to my bosses... :-)
SNIP
Glad I could help, that is why I posted the message to the
mmm... I just wondered, since it's very likely that most people ended up
deleting it *because* of the subject line. .. so it probably wont help ...
well it might...
I don't know -- It seems that plain English words are not in spam at all these
days... It would have read L AGR3 B*REAs3T5 or
On Fri, 2004-04-02 at 16:01, Martin Mielke wrote:
Hi all,
I installed all needed RPMs by GnoPhone to be installed without problems
but when attempting to install GnoPhone itself I get this message:
# rpm -Uvh gnophone-0.2.4-1.i386.rpm
error: Failed dependencies:
mozilla = 0.9.2
Nicolas Gudino wrote:
http://sip.house.com.ar/operator
Hi Nicholas,
Agree with the other feedback - looks beautiful, the auto-refreshes are
exceedingly smooth...definitely vindicates using Flash for client-side :)
I also agree that more buttons would be very useful. (Although some of
my labels
On Fri, 2004-04-02 at 16:51, Dave Tipton wrote:
Dave Tipton
Infrastructure Architect
817-858-9841 Voice
Euless, TX
Ham Radio Call Sign: W3DMT
--
The definition of insanity is doing the same
Get an RMA. I've had a few that did that as well.
Stephen Dolloff
DLS Internet Services
847-854-4799 x256
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Christopher J. Wolff
Sent: Thursday, April 01, 2004 5:50 PM
Very Nice!
I'm experiencing a bit of troubles in using for some kind of channels.
Actually it shows correctly the status only on ZAP/## channels, while i
can't see anything happening on SIP/ channels neither on IAX2/ channels
(neither with the new .pl you posted).
Regards,
--
Stefano Finetti
Gavin Hamill wrote:
I'm using Mozilla 1.7a installed from a tarball. The needed libraries
are just there:
You've answered your own question. You installed Mozilla from a tarball. RPM
therefore doesn't know about it. You need to install a recent Mozilla RPM :)
or use --nodeps
F
Andrew Kohlsmith wrote:
mmm... I just wondered, since it's very likely that most people ended up
deleting it *because* of the subject line. .. so it probably wont help ...
well it might...
I don't know -- It seems that plain English words are not in spam at all these
days... It would have
I'm having a problem configuring asterisk to send incoming calls to
Firefly.I can make outgoing calls from firefly through asterisk
without any problems at all. The firefly client does this when it's on
the same IP subnet without a firewall, or from a NAT'd environment. Can
anyone tell me
http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20vmail.cgi
Or did you mean asynchronously?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher
Lewis
Sent: Friday, April 02, 2004 6:27 AM
To: [EMAIL PROTECTED]
Subject:
I need to upgrade the kernel of my Redhat 7.3 (2.4.18-3) box because of a
bug. Does anyone know what kernel(s) can I use with asterisk-0.7.0,
libpri-0.5.0 and zaptel-0.8.0?
Thanks
Gary F.
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On Fri, Apr 02, 2004 at 08:52:09AM -0600, Eric Wieling wrote:
It's not asterisk, its the codecs. Codecs other than ulaw and alaw will
distort continuous tones like DTMF.
Welll ...
At work we experience this with Cisco dial-peers over G.729: DTMF is
erratic. But it's *NOT* inoperable.
Hi,
After a long way of problems (shipping, customs, etc) finally I got
Welltech working. Here below my comments.
- The documentation is poor and have errors
- The web configuration is not complete. However is useful for the basic
configuration parameters. The command line is necessary for
Andy Powell wrote:
1 Access to the PSTN - this can be done via a single X100P card (plugs into a standard phone line) or one of the sinlge port T1 cards or 4 port TDM410 cards (if you need a shedload of lines). You can also use a VoIP - PSTN gateway or gateway service (such as, but not limited to,
I don't know -- It seems that plain English words are not in spam at all
these days... It would have read L AGR3 B*REAs3T5 or something..
You mean like Best Web Hosting Service or Get Office Space Quotes ? :-)
I don't get spam like that.. .it's all misspelled or intentionally obfuscated.
Hello,
We are trying to migrate from an old application based on VOS to some
linux based telephony server. We are investigating bayonne and asterisk,
and we still don't know what is the best option for us.
One of the limitations is our old hardware, we have in stock some old
Dialogic boards.
I have a Welltech 3502 (2 FXS ports) and callerid will not work in SIP
mode. I contacted Welltech support and they informed me that callerid is
only working with the H.323 firmware. Once I flashed it with the H.323
firmware and figured out how to get it to work with asterisk, callerid did
indeed
On 02/04/2004 at 11:17 John Chambers wrote:
Andy Powell wrote:
1 Access to the PSTN - this can be done via a single X100P card (plugs
into a standard phone line) or one of the sinlge port T1 cards or 4 port
TDM410 cards (if you need a shedload of lines). You can also use a VoIP -
PSTN gateway
How would one hack the voicemail app to play saved vm messages back in a
'most recent first' fashion ? What source file is this defined in ?
Thanks,
Chris Clifton
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Hi
I´m a new user and I do test with my hardware.
I have a
x100p and telephone vozip.
And when I run this command asterisk
c for to test it.
My computer show it warning
[chan_iax.so]
= (Inter Asterisk eXchange)
== Manager registered action IAX1peers
== Parsing
Any ideas..???
[EMAIL PROTECTED]:/etc# modprobe wcfxs
/lib/modules/2.4.24-xfs/misc/wcfxs.o:
init_module: No such device
Hint: insmod
errors can be caused by incorrect module parameters, including invalid IO or
IRQ parameters.
You may find more information in syslog or the
vmail.cgi seems to be written in perl so modifying it should require
knowledge of perl and vi
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Clifton
Sent: Friday, April 02, 2004 10:51 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] voicemail
How
I have 2 ms messenger clients. I can not talk between them.
It shows them on-line on there PC. But on the contact list it shows them
not online. what can I do?
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We're having a problem with transfering calls. Our channels are not the
same as the extensions. We use words instead of numbers. So our config
looks like this:
SIP/HRUTTER,1,81101 Hildegard
SIP/JFOLEY-GS, 2,81103 Jerry
Consequently when I drag and drop to transfer a call
On Fri, 2004-04-02 at 10:12, Jim Rosenberg wrote:
On Fri, Apr 02, 2004 at 08:52:09AM -0600, Eric Wieling wrote:
It's not asterisk, its the codecs. Codecs other than ulaw and alaw will
distort continuous tones like DTMF.
Welll ...
At work we experience this with Cisco dial-peers
vozip wrote:
Hi
Im a new user and I do test with my hardware.
I have a x100p and telephone vozip.
And when I run this command asterisk c for to test it.
My computer show it warning
[chan_iax.so] = (Inter Asterisk eXchange)
== Manager registered action IAX1peers
== Parsing
Justin Carlson wrote:
vmail.cgi seems to be written in perl so modifying it should require
knowledge of perl and vi
The thing is, vmail.cgi isn't the voicemail application.
I've forgotten the password to my * box now so I can't look it up for you.
Look under asterisk/apps for app_voicemail.c
Hi,
On Thu, 2004-04-01 at 15:37, John Todd wrote:
At 9:35 AM -0500 3/31/04, [EMAIL PROTECTED] wrote:
Hi Yawl,
I took delivery this morning of a used BetaBrite LED
display sign which I promptly set about playing with.
Having found a windows app that grabs XML headline
files from places
How can do it.???
Where i can find it.?
Cheers.!
Vozip
-Original Message-
From: Anton Tinchev [mailto:[EMAIL PROTECTED]
Sent: viernes, 02 de abril de 2004 20:32
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] error with asterisk -c
vozip wrote:
Hi
I´m a new user and I do
Justin Carlson wrote:
just type it in it will remain until you restart your browser. ( it does
not disappear and you do not have to hit enter or anything like that)
I cut and pasted it right from the source code file, but no matter what
I do, I get the following line in debug:
La clave no
Tony Buser wrote:
I looked through your code to see if I could make some changes,
unfortunatly I can't speak Italian! :)
Not that unfortunate; the comments are all in Spanish, not Italian :-)
B.
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I don't want to re-invent the wheel if someone has already hacked a way
to do this.
One of my customers has a number of stores, and he wants to leave one
voicemail that would be delivered to all the managers at once. Each has
a voicemail account on his server.
I have googled around and
Hi Tony,
On Fri, 2004-04-02 at 14:13, Tony Buser wrote:
We're having a problem with transfering calls. Our channels are not the
same as the extensions. We use words instead of numbers. So our config
looks like this:
SIP/HRUTTER,1,81101 Hildegard
SIP/JFOLEY-GS, 2,
Hi all;
I am planning a PBX/Voice mail system for a small business (approx 12
employees with phones). They have an inbound ISDN PRI, which is
probably irrelevant because all inbound calls are routed first to
receptionists which rarely route the calls on (client is a medical clinic).
Any idea
Hi;
Sorry, I resent a message similar to the parent by mistake.
Best Wishes,
Chris Travers
Metatron Technology Consulting
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Does anyone have the physical spec sheet for the T100P from Digium? The one
on the website doesn't have what I need. Things like 3.3 or 5v operation,
uses n IRQ channels, requires 32-bit PCI, must be installed while standing
on one foot and reciting the GPL, etc. Also, if anyone is selling a used
Does anyone have the physical spec sheet for the T100P from Digium? The one
on the website doesn't have what I need. Things like 3.3 or 5v operation,
uses n IRQ channels, requires 32-bit PCI, must be installed while standing
on one foot and reciting the GPL, etc. Also, if anyone is selling a
On Thu, 2004-04-01 at 17:32, Scott Stingel wrote:
Hello-
Has anyone had experience connecting to a Marconi switch (in the UK) using
E1-PRI connections (TE410P)? In a new installation, my customer is getting
yellow alarms on every channel about every 30 seconds. These alarms clear
by the way, when I start up op_server.pl I get the following, even
though everything appears to work ok.
Use of uninitialized value in transliteration (tr///) at ./op_server.pl
line 67, CONFIG line 35.
Use of uninitialized value in string at ./op_server.pl line 68, CONFIG
line 35.
Use of
Hi,
On Fri, 2004-04-02 at 16:09, Tony Buser wrote:
by the way, when I start up op_server.pl I get the following, even
though everything appears to work ok.
Use of uninitialized value in transliteration (tr///) at ./op_server.pl
line 67, CONFIG line 35.
Use of uninitialized value in
Paul Mahler [EMAIL PROTECTED] wrote:
I have one SIP extension that can't logon to voicemail. The log file says
-- Incorrect password '3213' for user '4035' (context=other)
even though the context in voicemail.cnf says
4035 = 3213,Bill Smith
Did you solve this yet? Maybe you have
Ah, yes that line was a blank line.
Nicolas Gudino wrote:
Try removing line 35 on your op_server.cfg, maybe its a blank line and
the server does not handle that gracefuly. Its not harmfull anyways.
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Does anyone know if avaya voip product is running linux under the hood?
Thanks,
/glen
--
Glen Ford
[EMAIL PROTECTED]
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I heard it once that the Avaya's Definity runs linux but I am not
familiar with the product so sorry if it was wrong.
Lisa
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Glen Ford
Sent: Friday, April 02, 2004 2:48 PM
To: [EMAIL PROTECTED]
Subject:
On Fri, 2 Apr 2004, Glen Ford wrote:
Does anyone know if avaya voip product is running linux under the hood?
...
Probably not. Linux is GPLed.
More likely a propietary RTOS that they wrote themselves.
Tom
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FYI.
http://www.nwfusion.com/news/2003/1208avaya.html
New products on tap from Avaya include:
* The S8500 Media Server, a Linux-based call processor that supports up
to 3,200 phones.
Lisa
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lisa Xie
Sent:
Mark,
With CVS version are you using now?? is it working ok??
Luciano
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Mark
Messmore, Technical Support, University Telcom Inc.
Enviado el: Jueves 1 de Abril del 2004 10:38
Para: [EMAIL PROTECTED]
On Fri, 2004-04-02 at 14:00, Tom wrote:
On Fri, 2 Apr 2004, Glen Ford wrote:
Does anyone know if avaya voip product is running linux under the hood?
...
Probably not. Linux is GPLed.
More likely a propietary RTOS that they wrote themselves.
Sounds like you need to take a
Hi,
I am using Version .03, everything works fine except I can't
transfer by drag and drop. It seems to be a problem with flash since
the perl program is not outputting any debug info when I attempt
drag and drop.
--
Marvin Horst
Paul B Zimmerman, Inc
Nicolas Gudino wrote:
Version .03 is on
How would one hack the voicemail app to play saved vm messages back in a
'most recent first' fashion ? What source file is this defined in ?
apps/app_voicemail.c. Check vm_execmain() and the while loop at line 2866 or
thereabouts. The switch in there is the main voicemail menu (Press one to
Greetings,
I purchased a WiSIP at the VON conference and am now trying to configure it
to work with Asterisk. I have read all of the previous postings regarding
the WiSIP and most of the information apparently does not apply to the
version of firmware installed on my phone (version WF.00.0F).
I
- Original Message -
From: Steven Sokol [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, April 02, 2004 4:05 PM
Subject: [Asterisk-Users] WiSIP Firmware Version F?
I cannot get the WiSIP to register with my Asterisk box. It leases an IP
from my DHCP server, then immediately says
Hi list
I have configured some siemens optipoint 400 sip to work with asterisk.
I works very well with messages, moh etc... a good choice in my opinion...
Someone else have good/ bad experiences with that phones?
Miklos
___
Asterisk-Users mailing
I wan work * whith SIP Communicator, it is posible?, what is configurations?
who can helpme?
Thanks
Resgards, Jose
_
Charla con tus amigos en línea mediante MSN Messenger:
http://messenger.latam.msn.com/
Does anybody know of any commercial providers of IAX termination with
DIDs in the Seattle, WA area? I believe the area codes are:
425, 206, 253
Failing any commercial providers, is there anybody in the seattle area
running Asterisk with a PRI coming in who might be willing to sell me an IAX
Send the phone to me and let me have a play :-)
Steven Sokol wrote:
Greetings,
I purchased a WiSIP at the VON conference and am now trying to configure it
to work with Asterisk. I have read all of the previous postings regarding
the WiSIP and most of the information apparently does not apply
Hello-
I'm obviously doing something wrong here in trying to get an inbound
DID to work with voicepulse.
I have an outbound context set-up for those calls in iax.conf, and the
appropriate register in- statement.
within extensions.conf I am doing something like this:
exten =
On Apr 2, 2004, at 2:46 PM, Muiz Motani wrote:
Does anybody know of any commercial providers of IAX termination with
DIDs in the Seattle, WA area? I believe the area codes are:
425, 206, 253
Failing any commercial providers, is there anybody in the seattle area
running Asterisk with a PRI coming
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