Hi,
> -Original Message-
> > It could be deceiving indeed but the echo is slower than
> > usually. I did downgrade to asterisk 1.0.5 and the echo is
> less but still
> > there. I did not downgrade zaptel as I think it's not
> related. Or could it
> > be a zaptel timing problem?
As s
Gustavo García wrote:
Asterisk & SER are (more or less) SIP servers and Jabber is a completly
different protocol. There are no relation between jabber and Asterisk/SER.
There is a patch that adds Jabber notification to Asterisk in the bug
tracker.
Asterisk is not oriented to instant messaging
On Tue, 22 Mar 2005 03:51:15 +, Scheda <[EMAIL PROTECTED]> wrote:
> Hey, this is happening to anyone who I try this with. We get into the
> DISA, then hear the dial tone. Dial 1 then start dialing the number,
> and it hangs up. I thought adding a wait time after the DISA may help,
> I was wrong
After downgrading to CVS stable on Gentoo from *-1.0.5 my DTMF is not
working.
When I call-in and dial an extension phone is not ringing, same is with
password for my mail box is not recognized.
--
#Joseph
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Sys Admin wrote:
After 20 posts, in 2005 the ideal setup for a new installtion of a 50
user asterisk is:
Option1: IAX2 with softphone firefly
Option2: SIP with softphone
Option3: IAX2 with hardphones (which brand?)
Option4: SIP with hardphones.
Seems like we cannot come to a definite conclusion,
> All values for rxflash above 104 cause de "R" key presses to be ignored, and
> below 104 it's the same as if I hung up the phone. I think I must increase
> the hang up detection time value (?) somewhere so that 100ms hang ups are
> actually interpreted as flash hooks, but I haven't been able to f
After 20 posts, in 2005 the ideal setup for a new installtion of a 50
user asterisk is:
Option1: IAX2 with softphone firefly
Option2: SIP with softphone
Option3: IAX2 with hardphones (which brand?)
Option4: SIP with hardphones.
Seems like we cannot come to a definite conclusion, poll ?
so the v
Hello,
Does anyone know which EICON card is best with Asterisk in the USA?
There are a few, but I was wondering should I get an eicon diva 2.0, or
and eicon diva pro?
Thanks,
Greg
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On Mon, 2005-03-21 at 21:16 -0700, Tim Chandler wrote:
> Let me further clarify this. I am looking to buy the TE110P. The
> website says that "The TE110P with Asterisk will route voice and data
> traffic, and eliminate the need for an external router." How does
> this work? How is the data tran
I have three different time displays:
Flash panelcaller 615 48:00
called 62058:18
Snom phone shows for the same call 47:55
Why is there a difference at all?
bye
Ronald
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On Mon, Mar 21, 2005 at 01:03:52PM -0700, Kevin P. Fleming arranged a set of
bits into the following:
> Remco Barende wrote:
>
> >Are you sure? This is in the makefile:
> >
> ># Asterisk version, currently only v1_0 and HEAD are supported
> >ASTERISK_VERSION=v1_0
>
> Well, then the code is buggy
Hi Everyone.
Asterisk is one of those applications that need to be built from cvs on a
regular basis to keep up with the changes. I have always used package
management tools like apt.
How does everyone manage their Asterisk servers?
Geoff
___
Asteris
Hello,
I have come up with a php script that I believe should be able to send sms
alerts to cell phones as specified. I have added an option under the
extension account settings (am using [EMAIL PROTECTED] 0.6) successfully.
I would like to embed this script into the various specific configuratio
> I removed everything but what I needed to get the fax and email it to
> myself. So this is all I have, Thanks..
Well, then you need to renumber the priorities! Below is a start. Your
code does not do any emailing, by the way... what happens when you set
the EMAILADDR variable? Nothing... you nee
Tim Chandler wrote:
Let me further clarify this. I am looking to buy the TE110P. The website says that "The TE110P with Asterisk will route voice and data traffic, and eliminate the need for an external router." How does this work? How is the data transferred - as a pass-through like a NAT to t
I have the same motherboard. I put the card in the 2nd slot from the
bottom. In this slot, if you look at the manual, it will possibly be in
conflict with some USB channels. I believe I may have disabled one of
them but in any event I'm not using any USB devices. Otherwise, I
didn't have to
Let me further clarify this. I am looking to buy the TE110P. The website says
that "The TE110P with Asterisk will route voice and data traffic, and eliminate
the need for an external router." How does this work? How is the data
transferred - as a pass-through like a NAT to the server's netwo
I removed everything but what I needed to get the fax and email it to
myself. So this is all I have, Thanks..
[macro-faxreceive]
exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten => s,7,rxfax(${FAXFILE})
exten => s,103,SetVar([EMAIL PROTECTED])
exten => s,104,Goto(7)
Chr
Hi Everyone,
Thanks for all the input you add to the list. This seems to be a very good
list.
I am still new to Asterisk. If I run a PRI integrated T1 line into my office,
do I need to split the line between the data and voice before plugging it into
the asterisk box or is there some other w
Thanks I didn't see it. Sound like [EMAIL PROTECTED] isn't well liked on
this list..
Thanks for your help, David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dean
collins
Sent: Monday, March 21, 2005 7:39 PM
To: Asterisk Users Mailing List - Non-Commer
Hey, this is happening to anyone who I try this with. We get into the
DISA, then hear the dial tone. Dial 1 then start dialing the number,
and it hangs up. I thought adding a wait time after the DISA may help,
I was wrong. Here is what I have thus far in the DISA extentions.
[DISA]
exten => 7,1,DI
Hi Scott,
Intresting to know, cheers :)
Only down side though, is that most people using softphones will be
using Windows...
If only ALSA was available for windows ;)
-Shaun
Scott Williamson wrote:
That should be the program alsamixer, not amixer. Make sure to press F5
to get all of the playback/
Already available using group-settings in AMP.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Shaw
Sent: Monday, March 21, 2005 9:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] [EMAIL PROTECTED] .6 addin
That should be the program alsamixer, not amixer. Make sure to press F5
to get all of the playback/capture devices shown.
On Mon, 2005-21-03 at 22:02 -0500, Scott Williamson wrote:
> Ah, Console sound card echo.
>
> I found that with my cheep YMFPCI sound card that there is a channel
> called "wa
Ah, Console sound card echo.
I found that with my cheep YMFPCI sound card that there is a channel
called "wave capture" that is enabled for recording by default. And this
is only visible when one uses ALSA sound drivers. One needs to use an
ALSA mixer control program (I use amixer, the text mode o
Scott Bussinger wrote:
We just tried to go entirely with softphones in our office gave up after a
month or so of trying. I tried probably 10 different softphones running on
3.0Ghz WinXP machines and none of them were workable. I tried both SIP and
IAX2 softphones using headsets plugged into the aud
Nice, gUys...
Don't pick on Raphael too much. It's an honest mistake. I remember
making some of those -- although not of this caliber -- when I first
learned English a few years back. You really feel stupid once you
realize it... and I don't think the hugs had an implied meaning
either... just a f
Christopher Jacob wrote:
Would it be possible to put a T1 card in an asterisk box and use it to
simulate a PRI from the CO? As I build Asterisk boxes for customers I would
like to test the installation before getting on site.
Sure, Asterisk can act as the "network" end of a PRI as well, so you
cou
This sounds like an asterisk @ home issue. Not an asterisk issue.
Asterisk at home uses a GUI that limits what asterisk can do, look at the
config files it creates in (/etc/asterisk) and voip-info.org
Michael
On Mon, 21 Mar 2005, David Shaw wrote:
Hello, I tried to add an outside number (my cel
Hey All,
Would it be possible to put a T1 card in an asterisk box and use it to
simulate a PRI from the CO? As I build Asterisk boxes for customers I would
like to test the installation before getting on site.
If this will not work, can anyone share any other ideas for building (or
buying) asteri
Hello, I tried to add an outside number (my cell phone) to the group. I
would like to have both extensions ring as well as my cell. I'm running
[EMAIL PROTECTED] V.6.
Thanks, David
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I have created a call file which has been moved into the outgoing directory.
However the log file displays the following message: Unable to open
/var/spool/asterisk/outgoing/1.call: Permission denied, deleting
I have executed chmod 777 1.call on the file prior to moving it to the
outgoing direc
Hey Guys.
Im trying to make a script but I need some ideas on the logic behind it.
Here is what Im trying to do:
I have 2 zap channels. I want to make a macro that would do some random
choices on which of the 2 zaps to use, then test if it is available, if it
is, make the call, if not, then go to
I'm still not getting my outbound to work. I've seen two patches
relevant to broadvoice for chan_sip.c which apparently have already been
added to CVS. I'm dropping all outgoing calls after ~30 secs. Asterisk
doesn't seem to know they're gone though. I called my cell w/
broadvoice and turned on
Asterisk & SER are (more or less) SIP
servers and Jabber is a completly different protocol. There are
no relation between jabber and Asterisk/SER.
Jabber provides Instant Messaging and Presence and SIP also
provide that funtionality and includes general session management (for example
fo
Hi all,
I was wondering if someone could explain the relationship of
Asterisk, SER & Jabber (XMPP) to me.
I understand that there are facilities within Asterisk to
use jabber to notify of incoming calls via XMPP clients, however I’m
trying to work out exactly where the SER server wo
Sorry guys, i really made a mistake. I am so sorry.
thanks to help me, but i am reading the documentations in these sites.
Regards.
-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nome de Wiley Siler
Enviada em: segunda-feira, 21 de março de 2005 21:18
Para: Aster
What a terribly unfortunate way to begin and end a post. Even though
you technically aren't addressing the presumed majority of straight
asterisk users, let me answer:
Asterisk comes with plenty of sample configurations. Do a "make
samples" from your source directory to start off with something
What a great way to end the day! This one has me laughing my ass off
I am hoping you actually meant "guys". You may want to look up the meaning of
the word you used...
Here are the resources you should look into...
Make samples at the command line will create your asterisk sample scripts.
Hi Gays,
I beginning in the asterisk´s world. I need samples with simple
configurations to manager same sip connections only.
Please, i need help. are anybody help me?
one I hug to all.
Raphael C V Dannecker
Project Manager / Brasil SP
www.solutionsservicos.com.br
--
No virus found i
I've noticed that if I load the alsa drivers, I can't transfer from the
console.. Is that normal? It says the app is loaded, but when I type it
in, it says there's no such application 'transfer'.
I typically leave my console up on a transparent shell on my desktop and
use that instead of a sip or
For a easier comprehension, nowadays, H323 is like english. SIP is like
spanish and IAX is esperanto.
You can IAX. It's wonderful, modern, lot of advantages, pass through any
firewall, blah...blah..blah... but you can find only some strange guys
using that. :-)
Isamar
__
Thanks for the reply.
Yeah, I'm doing just fine without zaptel or libpri complied. And
while I could probably be fine as is regarding MOH, I'm getting the
constant notice from res_musiconhold.c:463 "monmp3thread: Request to
schedule in the past?!?"
I dug up some messages from this list from a fe
On 21 Mar 2005, at 21:53, Geoff Nordli wrote:
I am considering a G5/XServe for a conferencing system. I need to put
a TDM
card in the machine for timing. Is anyone out there using Asterisk on
Mac
with zaptel drivers? I am looking at using Linux for the OS. If so
what is
your experience?
Ha
[macro-faxreceive]
exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten => s,7,rxfax(${FAXFILE})
what happened to 2 through 6?
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sin
Hello All,
I am looking to get asterisk to receive faxes but I
really don't know where to go from here, any help pointing me in the right
direction would be great. What I would really like is for this second line
to answer faxes, but if a user typing in an extension it goes to that extensi
Hi again,
I have 2 of each of Polycom's SoundPoint IP300s, IP500s, and IP600s. I
got the 500s working first then later got the other phones.
My problem is that the current app version and bootrom on the 600s and
300s is 1.3.0.0711 and 2.5.0.0006 respectively and they don't seem to
want to upgr
Correct me if I did not understand it right, that would allow
restriction of who could monitor zap channels, but not which ones they
could monitor?
I need to be able to monitor only zap channels that are being used by
agents that belong to a specific group or queue, are you thinking this
can be d
Zach Scott wrote:
> Hey all again,
>
> I have successfully compiled and am running Asterisk (stable release)
> on OS X (10.3). However, any make directive that uses the "-shared"
> option in gcc results in an error. Apple states that -shared is not
> supported under OS X. Is there a workaround o
Look at bkw's valet parking
-d
At 03:58 PM 3/21/2005, you wrote:
I'm working through my list of features people will expect, and Hold
Pickup is at the top at the moment -- has anyone done any work on
this? We've had some unpleasant experiences with call parking, and
everyone seems to like the H
I'm working through my list of features people will expect, and Hold
Pickup is at the top at the moment -- has anyone done any work on this?
We've had some unpleasant experiences with call parking, and everyone
seems to like the Hold Pickup model. If you don't know what I mean by
Hold Pickup,
If your IP address changes daily, you need to continually update this, correct?
I too have this issue with SIP, but my ISP is a little unstable, so I
need to use DynDNS or No-IP to kinda work around/with it. I assume you
can't specify a hostname there?
--
Dana
On Mon, 21 Mar 2005 13:45:44 -0700
Somebody on IRC had one and said they were the greatest.
-Nate
Has anyone on-list had a chance to test these with * recently? I keep asking
about once a month hoping that someone has moved forward in using this
device. They look like great hardware, but early reports indicated that the
firmware
I've used the exact method MF describes here in production, works great,
and should even work if you move to an ARA environment.
-mike
MF Hulber wrote:
It seems the simplest approach is to create an extension(s) with the
password(s) and then if the incoming caller gets it correct, jump them
to
Hey all again,
I have successfully compiled and am running Asterisk (stable release)
on OS X (10.3). However, any make directive that uses the "-shared"
option in gcc results in an error. Apple states that -shared is not
supported under OS X. Is there a workaround or do I have just have to
live
> I'm testing a softphone-only setup (SJPhone with Plantronics
> 80 Headsets plugged into Soundcard) with around 40 users for
> that are linked over LAN in an organization of around 300
> people and never had any of the problems you described (the
> test is going for over a month now).
I'm glad it
Does anyone here have any experience with the Micronet SP5001 ATA and
Asterisk ?
Some "sip.conf" samples will help a lot.
Thore
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To
JB Wrote:
> For what it's worth, here's the chipset (tiger320):
>
>
http://www.tjnet.com/software/download/data_sheets/Tiger320_data_sheet.pdf
The data sheet for the chip(set) says: 3.3V power supply
supports 5V and 3.3V signaling.
Since there are reports that some of them (Original Digiu
> Did you consider vonage?
I played with Packet8 as a reality check before investing the time & money
into the Asterisk solution. I certainly wouldn't use Packet8 or Vonage or
any other service instead of an Asterisk solution. It's easy to get
termination delivered by IAX now and is both cheaper a
It seems the simplest approach is to create an extension(s) with the
password(s) and then if the incoming caller gets it correct, jump them
to another context. Otherwise stay in the default context and give
default prompts. I suppose you can find a way to read the password from
a DB or file o
Hi Mike,
I will test it again, and let you know.
Thorben
> -Oprindelig meddelelse-
> Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] På vegne af Mike Holloway
> Sendt: 21. marts 2005 21:49
> Til: Asterisk-Users@lists.digium.com
> Emne: [Asterisk-Users] IPSwitchBoard B
On Mon, 21 Mar 2005, Tom wrote:
> Quoting Peter Svensson <[EMAIL PROTECTED]>:
> > Also, you may not notice if you miss a ms worth of audio data, but the
> > digital signalling on a pri will. Ideally this should not be a problem but
> > with standard kernels it will be.
>
> This is what I have sus
Has anyone on-list had a chance to test these with * recently? I keep
asking about once a month hoping that someone has moved forward in
using this device. They look like great hardware, but early reports
indicated that the firmware questionable. Still, that was 6 months ago.
Michael
--
Michael
DISA?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josh Alberts
Sent: Lunes, 21 de Marzo de 2005 01:55 p.m.
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Script to Authenticate User and Dial Out
Hello. I'm looking for a script that I
What is your CLI output?
On Mon, 21 Mar 2005 15:03:14 -0400, Francisco Moreno
<[EMAIL PROTECTED]> wrote:
> Hello everyone...
>
> I'm trying to get up a testing pbx installation. Following instructions
> of what've read from the handbook and from asterisk's wiki, I wrote the
> dial plan as follow
Had the same problem:
I went into manager.conf and created a new user just for this purpose. You
should be able to use the information in the file to create your own. I then
created a "monitor" extension and combined, that was the information I put
into IPSwitchBoard.
-Kerry
-Original Messa
Hello. I'm looking for a script that I can use that will ask users for
a password, and then let them call any extension on the asterisk server.
Does anyone know of a premade script that can do this, or a resource I
could use to make my own?
Thanks,
Josh
__
I am considering a G5/XServe for a conferencing system. I need to put a TDM
card in the machine for timing. Is anyone out there using Asterisk on Mac
with zaptel drivers? I am looking at using Linux for the OS. If so what is
your experience?
Have a great day!
Geoff
__
Hey everyone,
I'm an * newbie and I'm trying to get things going here, and please
forgive me if you've heard this problem before. I have my * server on
OS X behind a NAT, and I'm trying to connect to SIP soft phones
(X-Lite) on various computers, some behind the same NAT and others
with public ad
Thorben,
I just installed the Microsoft .Net 2.0 beta x86 and IPSB 0.67.0.0 (on
windows XP), after filling in the config tab and clicking Connect to
Server, I get the following error:
Timer2: Index and length must refer to a location within the string.
Parameter name: length at System.String.Su
i had a similar problem a while ago. I solved it by defining
externip=xxx.xxx.xxx.xxx in sip.conf. It tells the remote SIP
client where you are.
-chuks.
Original Message Subject:
[Asterisk-Users] Can't hear the callerFrom: Lane
<[EMAIL PROTECTED]>Date: Mon, March 21, 2005 11:5
Comedian is probably a play on 'Meridian Mail' by Nortel.
It makes for a great laugh when you drop in a replacment for the Nortel
VM system with Asterisk.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dana Olson
Sent: Monday, March 21, 2005 12:51 PM
I am using Sprint for a couple of reasons.
First, they are a tear 1 internet provider.
Second, they are a tear 1 long distance provider.
Third, They are a CLEC in most markets in the US.
Fourth, They are the only Telco/Intenet provider not rocked by recent
Bankruptcy, Bad Customer Service or oth
On Mon, Mar 21, 2005 at 08:57:14AM -0700, Tom wrote:
> Quoting Tzafrir Cohen <[EMAIL PROTECTED]>:
> > You want to run a full desktop just be able to manage the Asterisk box?
> > That's what ssh is for.
> >
> > Xorcom Rapid added a menu application for managing the box for those who
> > don't know
> I've got a strange issue, that I haven't found addressed on the wiki.
>
> My asterisk box is behind a firewall which routes udp/tcp requests on 5060
> and
> 8000 to asterisk.
>
> When I make a call from a Zap or SIP extension on the inside of the firewall
> to any Zap or SIP extension on the
> -Original Message-
> From: Remco Barende [mailto:[EMAIL PROTECTED]
> Sent: Sunday, March 20, 2005 9:13 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Echo after upgrade * 1.05 -> 1.06
>
> >> The echo is quite slow, I would estimate about
Thats really strange.
Im testing a softphone-only setup (SJPhone with Plantronics 80 Headsets
plugged into Soundcard) with around 40 users for that are linked over
LAN in an organization of around 300 people and never had any of the
problems you described (the test is going for over a month now)
Hello.
I'm trying to transfer calls from an analog phone (Zap/1, TDM400P card) to
some other terminal connected to my Asterisk PBX. If I make a flash hook
pressing the phone hangup button quickly it works as expected, I get a new
dialtone and the other side is put on hold. But I would like to use m
I'm not having this issue... YES... it DOES stop when the user hangs
up... however, when I put someone back on hold (or go into a meetme
conference).. it will flush the buffer (takes about a minute or two)
and then once that has finished, it will begin playing the stream and
pickup right where it l
> The 4801 I'm doing is for a demo to hopefully get some funding. I REALLY
> need the fxo port. If I gave that up, what would be better? I GOT IT,
> *2* fxo ports!!
>
> So, I'll trade you for 2 fxo modules for the tdm400p.
My TDM400 has 4 fxo ports but i need them all :( I'm not running it in a
On Mon, Mar 21, 2005 at 09:09:16AM -0700, Tom wrote:
> Quoting Roger Gulbranson <[EMAIL PROTECTED]>:
>
> > On Mon, 2005-03-21 at 19:03 +1100, Adam Goryachev wrote:
> >
> > > In case you need it, there are X servers available for MS Windows
> > > platforms as well. Used to be one called exceed, but
On Sun, Mar 20, 2005 at 06:13:03PM +0100, Remco Barende wrote:
>
> >>The echo is quite slow, I would estimate about half a second
> >>or even more!
> >
> >Wow, that's enormous - However your ears can easily deceive you on this.
> >The
> >only way to know for sure is to record and analyse. Half a
I can cut and paste the log file from a reload right now, and provide
you with the other information when I get home after work:
tmp*CLI> sip debug
SIP Debugging Enabled
tmp*CLI> reload
Mar 21 14:52:42 NOTICE[23231]: indications.c:397
ast_unregister_indication_country: Removed default indication
Remco Barende wrote:
Are you sure? This is in the makefile:
# Asterisk version, currently only v1_0 and HEAD are supported
ASTERISK_VERSION=v1_0
Well, then the code is buggy, because the channel technology structure
stuff is only in HEAD, not 1.0.
___
As
pbx_dundi.c: In function `update_key':
pbx_dundi.c:1312: warning: implicit declaration of function `crc32'
pbx_dundi.c: In function `dundi_decrypt':
pbx_dundi.c:1368: warning: implicit declaration of function `uncompress'
pbx_dundi.c:1368: `Z_OK' undeclared (first use in this function)
pbx_dundi.c:
Ya got me thinkin'...
The 4801 I'm doing is for a demo to hopefully get some funding. I REALLY
need the fxo port. If I gave that up, what would be better? I GOT IT,
*2* fxo ports!!
So, I'll trade you for 2 fxo modules for the tdm400p.
Damn ... I'll have to build a new box for the 4801 ... Oh wel
Remco Barende wrote:
Now compiling chan_sccp.c 744 lines
chan_sccp.c:67: conflicting types for `sccp_request'
chan_sccp.h:1723: previous declaration of `sccp_request'
sccp_pbx.h:5: storage size of `sccp_tech' isn't known
make: *** [.tmp/chan_sccp.o] Error 1
That module is de
Remco Barende wrote:
Now compiling chan_sccp.c 744 lines
chan_sccp.c:67: conflicting types for `sccp_request'
chan_sccp.h:1723: previous declaration of `sccp_request'
sccp_pbx.h:5: storage size of `sccp_tech' isn't known
make: *** [.tmp/chan_sccp.o] Error 1
That module is designed
When I try to compile the new chan-sccp-easter2005 with asterisk 1.0.6 I
get the following error on make:
Now compiling chan_sccp.c 744 lines
chan_sccp.c:67: conflicting types for `sccp_request'
chan_sccp.h:1723: previous declaration of `sccp_request'
sccp_pbx.h:5: storage size
I'll give you 2 awesome X100P clones for one genuine Digium x100p 3.3
volt PCI soekris friendly card :)
On Mon, 2005-03-21 at 19:35 +, Senad Jordanovic wrote:
> [EMAIL PROTECTED] wrote:
> > Everything's for sale . if the price is right . STARTING BID?
> > :-)
>
> u start...
Hi,
I had the same problem, it is due to the fact that your server is
behind a nat, I solved the problem adding in my sip.conf
[general]
port = 5060
bindaddr=0.0.0.0
externip=200.121.56.70
localnet=192.168.1.0/24
context=llamadas
srvlookup=yes
Greetings
Carlos Rojas
Lima - Peru
On Mon, 21 Mar
On Mon, 21 Mar 2005 10:22:30 -0600, Brian Roy <[EMAIL PROTECTED]> wrote:
> Kerry,
>
> I'm more of a fan of anthm's patch that does this. You need to be
> running CVS-Head to get it though.
>
> http://bugs.digium.com/bug_view_page.php?bug_id=0002905
This is not available in 1.0.7?
aaron.glenn
__
[EMAIL PROTECTED] wrote:
> Everything's for sale . if the price is right . STARTING BID?
> :-)
u start... :)
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Everything's for sale . if the price is right . STARTING BID?
:-)
Senad Jordanovic wrote:
[EMAIL PROTECTED] wrote:
It works for me. The x100p (Digium 100 buck model) I have is slotted
for
3.3v and works fine.
My X100P cards are both clones and they do not work in my So
Dear Sirs,
Recently, I see a big interest for the web-meetme2!
Many of you have faced to different problems on compilation,
installation! I believe the approach to integrate DB into
meet was not the right way and with the recent functions
offered by the manager is really easy to get the same app
> Forget older years but in 2005 do hard phones really add any
> value over softphones.
>
> The call center agents already have p4 2.4ghz with 512 MB ram
> Win2K why not just get them a nice USB headset with a
> softphone IAX client,
We just tried to go entirely with softphones in our office gave
Tom wrote:
This is what I have suspected all along is that the signaling and timing
constraints on the PRI are such that you basically need asterisk running as a
real-time process. The whole point of the thread (in my mind) is if there is
anyway to cause X to not run as such a real-time process so
Hello,
I'm pretty new to asterisk (only been fighting with it on and off for
about the last month), so please go easy. I've been wrestling with the
documentation, forum posts, google, and my lack of telephony and VOIP
knowledge, trying to get my setup to work. My current problem has me
stumped s
Quoting Peter Svensson <[EMAIL PROTECTED]>:
> On Mon, 21 Mar 2005, Roger Gulbranson wrote:
>
> > On Mon, 2005-03-21 at 08:57 -0700, Tom wrote:
> >
> > > We don't want to have to spend an extra 3 grand for another
> > > server just to take up more space when we have this box that is sitting
> here
btw if there is no good softphone for IAX which does G729, i could
possibly get my company to buy consulting time from say DIAX and make
dante develop it further,
t
On Mon, 21 Mar 2005 11:03:48 -0800, Sys Admin <[EMAIL PROTECTED]> wrote:
> 2 reasons for not using IAX:
> A. CDR as part of the med
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