sure, we do that with a cron job that fires up a
script that connects to the GS web interface
and reboots is. the job is launched every 4 hours.
Also the GS web interface is down during a call,
so there's no risk to hangup undergoing calls.
(and the scripts also tries several times, before
going to
Thank You Steven.
Vijai.K
- Original Message -
From: "Steven Sokol" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, April 03, 2004 12:53 AM
Subject: RE: [Asterisk-Users] Asterisk Capacity
There are carriers using Asterisk to terminate thousands of lines. NuFone
has a data ce
Steven Sokol wrote:
There are carriers using Asterisk to terminate thousands of lines. NuFone
has a data center with 80 Asterisk servers in place. These installations
require a bit more engineering than the typical PBX server, but the system
does scale to extremely large systems.
Steven Sokol
Ow
Hi Azher,
> They advertised on TeleVision and then we had a rush of
> calls landing on the system (about 30 calls in 1-2 secs and 60 in 3-4
> seconds time).
this sounds like a high number of _simultaneous_ call attempts for a PRI
connected system to support. Your comments about a gradual load inc
Please have a look and give me your thoughts.
http://www.yottadot.com/callmanager/WAMi-alpha6.zip
Thanks,
Christian Hoffmeyer
YottaDot Solutions
Huntsville, AL
(iax) 700.859.4508
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Matthew:
> Does anyone know of an IAX/SIP DID provider in Vancouver, British
> Columbia? I'm looking for a voicepulse isk service, one DID with
> standard calling features and some sort of long distance package.
We have not announced yet but we have ordered our first PRI and redundant
hot-swap d
Hi,
My company is a call center and we are using * server for voip calls to america. * server is installed on a dual CPU machine and it is acting as a SIP/IAX2 gateway. SIP protocol is used for agents to connect to * server and * server used IAX2 protocol to connect to our VoIP service provider. T
There are carriers using Asterisk to terminate thousands of lines. NuFone
has a data center with 80 Asterisk servers in place. These installations
require a bit more engineering than the typical PBX server, but the system
does scale to extremely large systems.
Steven Sokol
Owner/Manager
Sokol &
Hi all,
I would like to know the real scalability of
Asterisk. Does anyone have any real numbers? What is the largest deployment of
Asterisk?
Thanks
Vijai.K
OK, I finally solved this by updating to the latest CVS snapshot.
Apparently there was some bug, that got resolved in the last 2 weeks since
I installed.
Thanks for the help.
--
Ken DeMaria
[EMAIL PROTECTED]
>
>>>I'm having a problem configuring asterisk to send incoming calls to
>>> Firefly.
Thanks for the suggestion. I had tried this, but other than the opvious
error, I don't know how to fix it. It appears to register just fine. But
when I call it I get a "cannot create channel of type IAX2".
Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ
Timestam
To all those that have been following this thread (and helping me) --
it is working.
The TN767 (refurb, but I just found out) was bad. Ah. Plugged into
another card that went to a Shiva dialup box -- and boom, right in.
So to all trying this, I know you can set this up in no time flat (as
Michael Graves wrote:
OK. So it would appear that my quest for FXO adapters unconvers more,
and certainly more mature, H.323 based devices...not so many SIP
devices. What would be the benefits of SIP over H.323 for a small
office * server? All I need to do is bring 4 POTS lines into * with
Caller
Does any one regularly reboot GS101? It sometimes lost registration with
* and needs to be reboot.
What is the best way to do it by cron?
David Kwok
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>
> Is your sip bound to an ip address on the server? If the Wisip were
> making
> it to Asterisk and failing to register you would know it because it would
> retry every 2 seconds.
I don't believe it is making it to Asterisk. I have tried changing the SIP
Proxy IP field to both the IP address
>> 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
I'm starting to get this to work! Well I got Voice Mail to work!
All calls goes to voice mail without ringing the users phone (iaxComm).
Here is my iax.conf and my extensions.conf
Any help would be great!!
Thanks
extensions.conf
Description: extensions.conf
iax.conf
Description: iax.conf
OK. So it would appear that my quest for FXO adapters unconvers more,
and certainly more mature, H.323 based devices...not so many SIP
devices. What would be the benefits of SIP over H.323 for a small
office * server? All I need to do is bring 4 POTS lines into * with
Caller ID, make outgoing local
Hi,
I can't seem to get any of my own MP3's to work with Asterisk as music
on hold. The default sample one works fine, but if I place another one
in there, Asterisk fails to start. I have removed all ID3 tag
information and also made it 128kbps mono. Is there something i'm
missing?
Thanks!
-
I have problems with talk off at times.. my wife, just can't stop her.
(sorry non asterisk related joke)
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, April 02, 2004 12:24 PM
Subject: Re: [Asterisk-Users] dtmfmode=inband with G.
On Apr 2, 2004, at 4:59 PM, Duane wrote:
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
Hrm, Jeremy told me it was.. but oh well - just send NuFone an e-mail or
Paypal them an amount with a user/pass and they'll set up your account ASAP.
- Joshua Colp.
- Original Message -
From: "Hermann Wecke" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, April 02, 2004 9:39 PM
On Fri, 2 Apr 2004, Joshua Colp wrote:
> http://join.nufone.net/ has some information, but it's under development.
> You can signup through there though.
After filling all the information, I got an error message:
"Something is seriously wrong
Column 'emailAddress' cannot be null"
Can someone con
JORA ROME wrote:
I wan work * whith SIP Communicator, it is posible?, what is
configurations? who can helpme?
Thanks
I couldn't get it to work either, circa early February. I had some
correspondence with the developer and sent him logs, etc. but nothing
ever came of it. He did say there were "a
Ken DeMaria wrote:
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. Ca
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 in who might be willing to
I've got it working with a PRI T-1. In the op_server.cfg just list each
channel as 'Zap/1, Zap/2', etc regardless of the fact that Asterisk
actually sees them as Zap/1-1, Zap/1-2, etc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Sent: Friday, April 02, 2004 7:52
Scott-
Thanks for the tip. It was in fact that I didn't have the two contexts
matching. Once I resolved that everything is working great.
-Steve
On Apr 2, 2004, at 6:21 PM, Scott Laird wrote:
On Apr 2, 2004, at 3:12 PM, Steven Kokinos wrote:
Hello-
I'm obviously doing something wrong here in
On Apr 2, 2004, at 3:35 PM, Joshua Colp wrote:
http://join.nufone.net/ has some information, but it's under
development.
You can signup through there though. If you really want opinions and
such,
you'll probably get them in this post. Personally I use NuFone for
toll-free
and outgoing, I have n
http://join.nufone.net/ has some information, but it's under development.
You can signup through there though. If you really want opinions and such,
you'll probably get them in this post. Personally I use NuFone for toll-free
and outgoing, I have never had a problem! It always seems to work.
- Jos
Their service is still considered to be in test or beta, but it works
fine. E-mail sales at nufone dot net and they will answer any questions
you have and set up an account if you like. They are currently on a
pre-paid basis, just PayPal them over what ever amount you want to start
with, and star
Nufone has an 800-number service? How did you find out about that.
I have looked at NuFone's website numerous times and I couldn't find any
details about the services they provide. Sure, they list general information
about their service, but no details such as which LATAs they provide DIDs in,
Hello,
I hate to ask here, but..
Does anyone know of an IAX/SIP DID provider in Vancouver, British
Columbia? I'm looking for a voicepulse isk service, one DID with
standard calling features and some sort of long distance package.
I've looked around on voip-info.org's list of VoIP providers bu
On Apr 2, 2004, at 3:12 PM, Steven Kokinos wrote:
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 doin
No one responded to the original, nor to my followup. So, here I am
again following up my own followup :-)
I was speaking with a colleague of mine today who is running * at his
office and at home. He told me that he was using iaxcomm and couldn't
hear any sounds. I told him that I had the same
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
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 => 212xxx,1,Dial
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 to
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
tr
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/
__
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
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- 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 immediate
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
Where do I find info on how to set up the auto-attendent??
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> 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
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 t
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
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]
Asunto
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:
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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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: [Aste
Does anyone know if avaya voip product is running linux under the hood?
Thanks,
/glen
--
Glen Ford
[EMAIL PROTECTED]
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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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"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? Mayb
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, line 35.
> Use of uninitialized value in string
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, line 35.
Use of uninitialized value in string at ./op_server.pl line 68,
line 35.
Use of uninitialized valu
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
> th
> 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
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
T
Hi;
Sorry, I resent a message similar to the parent by mistake.
Best Wishes,
Chris Travers
Metatron Technology Consulting
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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;
I am in the process of planning a PBX/Voice mail system for a business
with an ISDN phone system and somewhere on the order of 12 internal
phones. The ISDN system appears to be a Primary Rate Interface, though
this may be irrelevent because receptionists answer all inbound
telephone calls
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,
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 looked
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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http:
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 coinci
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 d
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 fro
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
vozip wrote:
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 '/etc/as
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 di
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 c
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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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 w
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
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
== Pars
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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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 g
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
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. Does
> >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 obfu
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,
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 modi
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.
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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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: [Asterisk-Users
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 whe
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 r
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
___
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
_
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
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 s
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 16:01, 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 >
> 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"
-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... :-)
<>>
Glad I could help, that is why I posted the message t
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 would
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 >=
Title: Message
I am just an
Asterisk newbie doing a test install. I am using 2 X-Lite clients and
have configured 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 th
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 s
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