Thanks for the reference, JonJon Radon <[EMAIL PROTECTED]> wrote:
This issue is well documented.http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20vmail.cgiOn Tue, 11 Jan 2005 04:12:53 -0800 (PST), Frank Kostin<[EMAIL PROTECTED]>wrote:> Hi, Just doing a "chmod" OK> > Halas, not a special
Any chance to post a small howto with the correct server settings et al?
Thanks!
Remco
On Wed, 12 Jan 2005, Wilson Pickett wrote:
Solved. Required peer and friend.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mai
Solved. Required peer and friend.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
> In some cases, the IAXy device and/or Asterisk are not communicating their
> qualification, because "iax2 show peers" shows the device as status UNKNOWN.
> However, when a user picks up the telephone plugged into the IAXy, they can
> place a call just fine within our Asterisk server.
Are the IA
Hello, I am running 1-11 CVS-HEAD.
Per the sip.conf.sample, I should be able to use a register line like:
register => username:[EMAIL PROTECTED]/ext if I have [sip_proxy]
defined below it.
I can't get it to work at all. I keep getting Jan 11 23:47:49
WARNING[1388]: chan_sip.c:1348 create_addr:
Well, I can't find a softphone thus far for linux that works with IAX.
I only have one computer running so far. But in a few weeks I will be
able to get another box to set asterisk up on and then I can use
windows as well. If anyone knows of a linux applicable IAX softphone,
I'd be more than willin
Which H323 module are you using? chan_h323 or oh323?
On Tue, 11 Jan 2005 22:55:02 -0600, Voip Business
<[EMAIL PROTECTED]> wrote:
> Hello List
>
> I am testing a new carrier with h323 (for a special need) but when I
> dial to a pstn phone this is what happens:
>
> both sides rings
>
> Pstn l
We are forming an Asterisk User Group in Winnipeg. Our first meeting
will be sometime in the last half of February.
If you are interested in participating please join our mailing list:
http://www.muug.mb.ca/mailman/listinfo/asterisk
Look forward to seeing you at our first meeting,
Bill
__
I just tested another way...
On windows, install Cygwin, download iaxyprov, make, and you can run
it under Cygwin
It works, I just provisioned my IAXy with it
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailma
On Tue, Jan 11, 2005 at 10:44:19PM -0300, Bartosz Jozwiak spake thusly:
> We are looking for commercial solution SS7 with Asterisk.
> It does not need to be "build-in" with Asterisk.
> Could anybody suggest something?
I see a lot of people asking for asterisk and ss7. Just what exactly do
these pe
I'm using CentOS - which is another Red Hat Enterprise clone, like WBEL
www.centos.org
I've had no problems of any kind with the OS
- Original Message -
From: "Imran Sadiq" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, January 11, 2005 9:58 PM
Subject: [Asterisk-Users] What is the best and easie
Red Hat ES is easy to use with lots of support and docs onthe www.
-Original Message-
From: <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Wed, 12 Jan 2005 00:00:18 -0500
Subject: Re: [Asterisk-Users] What is the best and easiest flavor to be
used wi
try this : http://www.whiteboxlinux.org/
On Wed, 12 Jan 2005 16:58:29 +1300, Imran Sadiq <[EMAIL PROTECTED]> wrote:
>
>
>
> Could anyone please advise me on the best flavor of Linux on which Asterisk
> is easiest to install.
>
>
>
> I am currently using RH8.0, everything over the IP wo
Hello List
I am testing a new carrier with h323 (for a special need) but when I
dial to a pstn phone this is what happens:
both sides rings
Pstn line no audio when pickup
ip phone continue ringing.
I am using h323 (Not oh323)
the carrier says is a slow start and need fast start ,, how is it
Rich Adamson wrote:
>>> I'm trying to connect a TDM400P with an FXS module to a Valcom
>>> V-9940 Paging adaptor. This port on the TDM400P was connected to a
>>> 2500 Set and was working I just re-connected it to the Valcom (which
>>> is known to work on a Telco POTS line) and its not picking up.
Could anyone please advise me on the best flavor of Linux on
which Asterisk is easiest to install.
I am currently using RH8.0, everything over the IP works
fine but when I want to call a physical line I can only have conversation for
about 3 sec and everything freezes after that.
I
At the time I didn't realize it was a common error. I thought it was a
problem with the Makefile. I promise to google before I post :)
Anyway it works now... somewhat. tiff's are incomplete but I will have
to troubleshoot more.
On Wed, 12 Jan 2005 14:37:48 +1300, Matt Riddell
<[EMAIL PROTECTED]>
On Tue, 2005-01-11 at 21:43 -0500, Michael Greb wrote:
> On Tue, Jan 11, 2005 at 10:47:27AM -0500, Andrei (MPI) wrote:
> > Sync up your clock, guys: on the * server, PCs and the phones. And the
> > problem will go away.
> >
> > Andrei
>
> I can assure you this will not fix the problem, all of my
Actually the problem was in the source code. I reinstalled the 1.0.3
sources from the tarball and rebuilt. I had to do this somewhat
manually as a checkout -r v1-0 from CVS wasn't downgrading the
chan_iax2.c file (which was changed yesterday) and many others.
Probably doing something wrong. Anyway
Hi Andrew,
E100P does work with Telstra, its not not Aticked. Its up to you if you
want to not tell them. E400 does.
- luke
Andrew Thrift wrote:
Hi All,
We are currently running Asterisk 1.0.3-BRISTUFFED on Gentoo Linux
with a 2.6 Kernel and this is connected to a Telecom ISDN-BRI
connection wi
Title: AMP Anyone?
Hi all, I have been using Asterisk for a while now, and loving it. Just about to update to 1.0 (running like 0.93)
I was wondering if anyone has any expertise in the implementation of AMP onto an existing Asterisk install? The instructions for it all deal with a fresh in
> > I'm trying to connect a TDM400P with an FXS module to a Valcom V-9940
> > Paging adaptor. This port on the TDM400P was connected to a 2500 Set
> > and was working I just re-connected it to the Valcom (which
> > is known to
> > work on a Telco POTS line) and its not picking up. The
> > Valcom
On Tue, Jan 11, 2005 at 10:47:27AM -0500, Andrei (MPI) wrote:
> Sync up your clock, guys: on the * server, PCs and the phones. And the
> problem will go away.
>
> Andrei
I can assure you this will not fix the problem, all of my asterisk
servers, and so far hard phones, have ntp setting the time.
I found this to be missing... thank you,
However after restarting, rebooting, and re-building.. no change..
Still no callerid on the Zap channels...
It is however working on my 7960 so I know the feed from the FXO card is
working fine..
I have even specified it not only in the Channels context but
On Tue, 2005-01-11 at 21:05 +0100, Vikram Rangnekar wrote:
> After 20 cups of strong coffee and wasteing most of tonight and obviously
> doing lots of googling and emailing many people, i've concluded that dlink
> voip phones specifically DPH-80 dosent work with asterisk.
>
> If anyone has had an
That was it. Thanks for the help.
B. J.
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 11, 2005 18:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] "o" extension broken?
check to see if you have "o
Scheda wrote:
Hey, I'm having some errors whenever I dial out and I can't dial in at
all. I'm using NuFone as my provider just so you know.
Jan 11 17:39:46 WARNING[1771]: chan_oss.c:413 soundcard_setinput:
Unable to re-open DSP device: No such device
Jan 11 17:39:46 WARNING[1771]: chan_oss.c:572 os
Brian Dingman wrote:
I grabbed the latest sources from CVS yesteday and am having problems
compiling. * v1.0.3 was running previously without issue. I tried
checking out the older source but get the same make errors.
The box is running RH 9. I am getting the following errors. Any
thoughts on what i
On Wed, 2005-01-12 at 12:40, Matt Riddell wrote:
> Ferguson, Michael wrote:
> > G'Day All,
> >
> > rpm -q kernel-source returns "Package kernel-source is not installed"
> > Where can I find it and install it. Asterisk evidently needs it for a
> > successful install.
>
> You can do:
>
> yum insta
Steve,
What version of MySQL are you running? I upgraded to 4.1.8 and ran into the
problem below. I initially tested with the user root and the default blank
password and was OK. But when I changed over to a new user with a password,
I noticed an error message in the httpd logs:
Client does no
Ferguson, Michael wrote:
G'Day All,
rpm -q kernel-source returns "Package kernel-source is not installed"
Where can I find it and install it. Asterisk evidently needs it for a
successful install.
You can do:
yum install kernel-source (although I thought you didn't need it in 2.6)
--
Cheers,
Matt Ri
Steve Underwood wrote:
The answer to this problem is the same as for every other time the same
question has been asked.
What he means is that if you put your error into google and click
search, it will come up with the same question being asked and answered
multiple times.
Congratulations on th
Hello,
We are looking for commercial solution SS7 with Asterisk.
It does not need to be "build-in" with Asterisk.
Could anybody suggest something?
Thank you in advance.
Bart
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digi
On Wed, 2005-01-12 at 00:47 +, Asterisk wrote:
> Comments inline...
>
>
> Christopher L. Wade wrote:
> > Comments inline...
> >
> > Asterisk wrote:
> >
> >> However, I feel that it would be more useful for the manager output to
> >> be tab delimited, one record per line:
> > Why? How?
>
Timing is probably for external timing sources, like gps receivers or in
telco arenas it's common to get timeing and have a clock slaved to it and
all channel banks are externally timed to that synced clock.
Lyle
- Original Message -
From: "Matthew Boehm" <[EMAIL PROTECTED]>
To:
Sent: T
Peter,
Should I do the with pri intense debug span or pri debug span only?
I will need a little more time for install again the T1's.
Alejandro
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/
Oh absolutely :)
Dante ... Don Quixote.
Perhaps if you are surrounded by dragons you should roast hotdogs and
marshmallows?
At any rate their will always be people using HTML, Text, and Test
messages and it's far easier to privately message them then to do battle
with the windmill of re
On Tue, 2005-01-11 at 19:45 -0500, C. Savinovich wrote:
> And what does that do?
>
> I am not testing no filter, I am testing the change of my name as it shows
> up on the list. I can't think of any other way. However, if turning off
> HTML has the desired result, I thank you for the tip.
>
You know, you have a choice.
C. Savinovich
ITN-Telecom
>>I don't know. Dealing with some people here seems like I am in hell
>>tilting at dragons.
On Tue, 2005-01-11 at 20:02 -0500, C. Savinovich wrote:
> Brian:
>
> Do you mean Don Quixote and the windmill?
>
> CS
>
>
> >>Dante
On Tue, 2005-01-11 at 20:02 -0500, C. Savinovich wrote:
> Brian:
>
> Do you mean Don Quixote and the windmill?
>
> CS
>
>
> >>Dante and the windmill?
I don't know. Dealing with some people here seems like I am in hell
tilting at dragons.
>
> -Original Message-
> From: [EMA
Brian:
Do you mean Don Quixote and the windmill?
CS
>>Dante and the windmill?
Brian Greul
Texas Shirt Company
www.txshirts.com
713-802-0369 / 713-861-6261 (fax)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, J
Comments inline.
Peter Svensson wrote:
On Tue, 11 Jan 2005, Asterisk wrote:
In my mind, (yes, a small one compared to the giants walking around
here) There are several advantages in this method:
a) Parsing one line of data per record is in order of magnitude easier
to code.
Not really, unless
Dante and the windmill?
Brian Greul
Texas Shirt Company
www.txshirts.com
713-802-0369 / 713-861-6261 (fax)
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 11, 2005 5:44 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial D
Comments inline...
Christopher L. Wade wrote:
Comments inline...
Asterisk wrote:
I am currently writing a prototype agent monitoring system, which (as
most others in question) simply monitors the output from the event
system, and displays the relevant information. I would hope to donate
this bac
On Wed, 2005-01-12 at 11:01, Matthew Boehm wrote:
> what is g723? ive never seen that before...
It's a codec. and it look like you have some form of codec translation
problem.
>
> -- Executing Answer("Zap/1-1", "") in new stack
> -- Accepting call from '2819870065' to '2815692780' on channel 0/1
And what does that do?
I am not testing no filter, I am testing the change of my name as it shows
up on the list. I can't think of any other way. However, if turning off
HTML has the desired result, I thank you for the tip.
BTW: I can only dedicate 5 minutes to this issue. Thanks
CS
On
How can I check it?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Tuesday, January 11, 2005 7:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] No sound for music on hold
You have to make sure you
You have to make sure you have a timer source.
On Tue, 11 Jan 2005 17:06:45 -0700, Mark <[EMAIL PROTECTED]> wrote:
> Did you build the symbolic link?
>
> ln -s /usr/local/bin/mpg123 /usr/bin/mpg123
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
/SNIP/
I cannot for the life of me get it to register with the asterisk server,
nor upgrade the firmware to the latest (1.41) i'm still using 1.37.
/SNIP/
You will never be able to upload the firmware - as long as you try to
upload the firmware in 'debug' mode. This is the safe process to make
s
I've tried it and didn't get it working, I also asked when calling
their support and they said chances are relatively high that they
won't let you do it if you ask nicely (to management) and that it is
for sure disabled by default.
mitchel
On Tue, 11 Jan 2005 18:50:41 -0500, digium-list <[EMAIL P
Did you build the symbolic link?
ln -s /usr/local/bin/mpg123 /usr/bin/mpg123
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vitalie Apostu
Sent: Tuesday, January 11, 2005 4:19 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] No sound for
check to see if you have "operator=yes" defined for each voicemail user.
see patch: http://bugs.digium.com/bug_view_page.php?bug_id=0003080
-matthew
- Original Message -
From: "B. J. Bomar" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Tuesda
what is g723? ive never seen that before...
-- Executing Answer("Zap/1-1", "") in new stack
-- Accepting call from '2819870065' to '2815692780' on channel 0/1, span 1
-- Executing Wait("Zap/1-1", "1") in new stack
-- Executing SetVar("Zap/1-1",
"FAXFILE=/var/spool/asterisk/fax/1105486770.492.tif")
Dose someone have been able to connect Micrrosofto Portrait Pocket PC
Version to asterisk?
Any information will be appreciate
Thanks
Erick
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asteris
The answer to this problem is the same as for every other time the same
question has been asked.
Steve
Brian Dingman wrote:
I edited the makefile and asterisk builds properly, but when I go to
start it, I get the following error:
[app_rxfax.so]Jan 11 18:44:12 WARNING[13877]: loader.c:258
ast_load
My question was about setting the caller ID to be something other than
the broadvoice phone number. That way if I want to put an 800 number on
their display I can do so. I also need this because I am setting up
incoming DID numbers to the pbx. I need to set caller ID to show the
correct callbac
I edited the makefile and asterisk builds properly, but when I go to
start it, I get the following error:
[app_rxfax.so]Jan 11 18:44:12 WARNING[13877]: loader.c:258
ast_load_resource: libspandsp.so.0: cannot open shared object file: No
such file or directory
Jan 11 18:44:12 WARNING[13877]: loader.
That firmware does NOT work. From what I can tell, with that firmware, when
you select IAX2, it actually uses the Net2Phone protocol. In other words,
the "IAX2" support doesn't exist, and is just an alias for N2P. Indeed, from
what I've seen, the phones will even say "N2P" when booting.
We have so
On Tue, 2005-01-11 at 18:30 -0500, C. Savinovich wrote:
> This is a test, please disregard
Next time you post, make sure you turn off HTML.
And not that I expect anything productive to come of this part of a
rant... If you(collective mass of people sending test messages lately)
are testing filter
It is defined, and was working on a previous version of Asterisk. I'm not
sure when it stopped working.
B. J.
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 11, 2005 16:55
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
> CREATE TABLE `cdrs` (
>
> `cardnum` char(40) default NULL,
>
> `callerid` char(80) default NULL,
>
> `callednum` char(80) default NULL,
>
> `trunk` char(40) default NULL,
>
> `disposition` char(20) default NULL,
>
> `billseconds` int(11) default NULL,
>
> `billcost` int(11) default NULL
> ) TYPE=
Same problem. I dropped all the tables and ran the above. I also checked
the sockets and they are correct. Any other ideas?
I am still stuck on this.
Allright, let us do it the easy way. Just copy and run the script below
on your
Mysql Command line. Lets see how this goes.
If you want to u
Greetings,
I just received some netweb-301 phones frm Seshu down in NJ.
I cannot for the life of me get it to register with the asterisk server,
nor upgrade the firmware to the latest (1.41) i'm still using 1.37.
The packets are traversing the router, going into the other subnet,
hitting the ast
This is a test,
please disregard
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-use
Warren Burstein wrote:
Michael Welter wrote that I should be worried about the usb module.
Would "rmmod uhci_hcd" be enough, or should I disable it in the BIOS
like Shoval said?
Also, after the rmmod, I still have the conflict with libata on 169
CPU0 CPU1
0:731100672525
Hello,
I am having a couple
of problems. Any help is appreciated.
1. The voice mail
messages arrive in the mailboxes but when I play them back, the IVR tells the
time and date of the message but never plays it. It is as if it skips it.
2. Asterisk never
seems to send the voice mail as
Greetings,
I try to set-up Music-on-hold. I use X100P.
[mainmenu]
exten => s,1,Answer
exten => s,2,mp3player(/var/lib/asterisk/mohmp3/fpm-calm-river.mp3)
File fpm-calm-river.mp3 exist but there is sound in line. If I play this
file using mp123 I can here sound in my sound-boxes but the
Comments inline...
Asterisk wrote:
I am currently writing a prototype agent monitoring system, which (as
most others in question) simply monitors the output from the event
system, and displays the relevant information. I would hope to donate
this back to the community once it works properly :)
Hi..
Can someone help
telling me how to enable a full debug mode and how to turn it off
again.
I need to see what
Asterisk is doing behind the scenes. I am able to see the SIP debug events only
now. But I still need to see things like voicemail to e-mail
activities.
Thanks
Walid
_
On Tue, 11 Jan 2005, Asterisk wrote:
> In my mind, (yes, a small one compared to the giants walking around
> here) There are several advantages in this method:
>
> a) Parsing one line of data per record is in order of magnitude easier
> to code.
Not really, unless you have to invent string han
Hi,
I'm in the need to use a line extender. I found a device, which
should best be placed in the middle of the line, which would make
power supplying difficult, unless I'd use its "telco power supply"
features:
The device is capable of taking current from one signal wire pair
or voltage from the po
Michael Welter wrote that I should be worried about the usb module.
Would "rmmod uhci_hcd" be enough, or should I disable it in the BIOS
like Shoval said?
Also, after the rmmod, I still have the conflict with libata on 169
CPU0 CPU1
0:73110067252568IO-APIC-edge tim
Outgoing caller-id seems to work fine. BroadVoice appears to send the
name that is on the account and the phone number. My dial plan uses
SetCallerID and SetCIDName, but the later is definitely ignored and
the former may not actually be required.
- |Daryll
_
On Tue, 11 Jan 2005, Alex G Robertson wrote:
> http://ckp.made-it.com/g704.html says:
> "G.704 is the framing specification for G.703. A carrier can 'steal' a 64kbps
> time slot (TS0) from a 2.048 Mbps line and use this to provide timing. The
> result is that 31 time slots are left for data, whi
I am currently writing a prototype agent monitoring system, which (as
most others in question) simply monitors the output from the event
system, and displays the relevant information. I would hope to donate
this back to the community once it works properly :)
However, I feel that it would be mo
you need to define the "o" extension to do whatever you want it to do.
-Matthew
- Original Message -
From: "B. J. Bomar" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Tuesday, January 11, 2005 4:45 PM
Subject: [Asterisk-Users] "o" extension br
Hey, I'm having some errors whenever I dial out and I can't dial in at
all. I'm using NuFone as my provider just so you know.
Jan 11 17:39:46 WARNING[1771]: chan_oss.c:413 soundcard_setinput:
Unable to re-open DSP device: No such device
Jan 11 17:39:46 WARNING[1771]: chan_oss.c:572 oss_write: Unab
Hi,
Il giorno mer, 12-01-2005 alle 00:38 +0200, Shoval Tomer ha scritto:
> Only if you don't have Digium hardware installed.
yes
> And only for MeetMe, I think.
>
> Correct me if I'm wrong on this, though...
really, it works for zaptel timing, that's needed only
by meetme and iax2 trunking. But
Hello all. I
just found out that I am no longer able to exit out of voicemail properly by
hitting the 0 key, but the * key works. Asterisk comes back and says "I'm
sorry, I did not understand that response" and goes on in the context. Is
this a new "feature" or bug? Is anyone else having
Only if you don't have Digium hardware installed.
And only for MeetMe, I think.
Correct me if I'm wrong on this, though...
> -Original Message-
> From: John Middleton [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, January 12, 2005 12:26 AM
> To: Asterisk Users Mailing List - Non-Commercial
Isn't this used as a timer source by zaptel?
On Wed, 12 Jan 2005 00:14:30 +0200, Shoval Tomer <[EMAIL PROTECTED]> wrote:
> You can disable the USB in the BIOS of the machine if you don't plan on
> using it.
>
> > -Original Message-
> > From: Michael Welter [mailto:[EMAIL PROTECTED]
> > Se
I am still stuck on this.
Allright, let us do it the easy way. Just copy and run the script below
on your
Mysql Command line. Lets see how this goes.
If you want to use webmin to find what the problem is go to webmin ->
servers ->Mysql and check the configuration there. My guess is that
there
I had a little billing problem once.. my credit card on file had expired,
and it took a bit of work to finally find somebody there who could get
their system to acknowledge my updated card number and quit sending me
"pay up or be disconnected!!" warnings.
My recommendation: call them on the phone.
You can disable the USB in the BIOS of the machine if you don't plan on
using it.
> -Original Message-
> From: Michael Welter [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, January 12, 2005 12:05 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users]
You absolutely do need to worry about usb module.
http://www.microsoft.com/whdc/system/sysperf/apic.mspx
Warren Burstein wrote:
I'm not having any trouble with interrupts, but here's my
/proc/interrupts on Fedora Core 2 on a hyper-threading CPU and using the
SMP kernel (2.6.5-1.138). I don't thi
It appears that it just supports SIP.
[EMAIL PROTECTED] wrote:
X-Ten is beta tesing a new softphone for Linux and could use some beta
testers. If interested, please contact: Neil McGuigan, [EMAIL PROTECTED]
I am currently testing it and it appears every bit as good as the
Windows version.
Will /
I grabbed the latest sources from CVS yesteday and am having problems
compiling. * v1.0.3 was running previously without issue. I tried
checking out the older source but get the same make errors.
The box is running RH 9. I am getting the following errors. Any
thoughts on what is wrong?
gcc -share
/SNIP/
-Original Message-
I am still stuck on this. How did you add more privileges? I have webmin
installed if its easier than command line. I am still getting the hang of
linux.
Solution: Follow these steps
-
This may not be due to privileges but due to the way database is cr
My question is simply, has anyone received a deposit from these people once
you return the equipment in good order? I've been unable to contact them now
for almost 2 whole months.
Thanks,
Bill Church
[EMAIL PROTECTED]
Bill,
Although this is quite OT, I'll reply.
I signed up for their service un
Hello all,
I tried out BroadVoice a few months back and got a WiSIP phone with it. The
service is spotty at best and after a few months I decided it really didn't
work well enough to keep around, although the WiSIP was nice.
I e-mailed their support and found out what I needed to do to cancel,
ap
/SNIP/
-Original Message-
I am still stuck on this. How did you add more privileges? I have webmin
installed if its easier than command line. I am still getting the hang of
linux.
Solution: Follow these steps
-
This may not be due to privileges but due to the way database is
Thanks very much
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Ishmael
Sent: Tuesday, January 11, 2005 3:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Installing * on fedora 3
Usually you select
Brian, thanks for your reply. I was under the impression that I need
two public IP addresses for the STUN server to work. Is this correct?
I only have one IP at the office now.
Thanks,
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Wilkins
I'm not having any trouble with interrupts, but here's my
/proc/interrupts on Fedora Core 2 on a hyper-threading CPU and using the
SMP kernel (2.6.5-1.138). I don't think I need to worry about uhci_hcd,
nothing is plugged into USB, but libata is the disk driver. How do I
get libata and wctdm
Yes, its me again!
I have found an issue with AgentCallBackLogin.
If there are calls in the queue waiting to be transferred to an agent.
And then I log into take calls via the AgentCallBackLogin The call in
the queue
Is transferred to me faster than I can enter in my password, and the
(extension
Usually you select to install the kernel during the installation of FC3, but
I think you can also do:
up2date --get-source kernel
Here's more info:
http://fedoraforum.org/forum/showthread.php?t=29315
Hope that helps,
Dave
-Original Message-
From: Ferguson, Michael [mailto:[EMAIL PROTEC
I am still stuck on this. How did you add more privileges? I have webmin
installed if its easier than command line. I am still getting the hang of
linux.
- Original Message -
From: "Rafael J. Risco G.V." <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Has anyone had any experience with the Planet VIP-101T H.323 Phone or
the VIP-150T SIP phone with Asterisk?
--
Regards,
Sean Milheim
iDREUS Corporation
--
This message was scanned for spam and viruses by BitDefender
For more information please visit
http://www.idreus.com/index.php?page=pr
G'Day All,
rpm -q kernel-source returns "Package kernel-source is not installed"
Where can I find it and install it. Asterisk evidently needs it for a
successful install.
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Ishmael
Sent: Tuesday,
Hello.
I have my * server set up and working perfectly. I wanted to allows
calls to sip:[EMAIL PROTECTED] In sip.conf, I have:
[general]
context=default
Also, in extensions.conf, I have:
[default]
exten => myname,1,Goto(internal,nabeel,1)
However, when I make a
1 - 100 of 223 matches
Mail list logo
