Dear David
I want customized packages to be installed from cd with no need every time
to install packages and my personalized web interface ,
Regards
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Gomillion
Sent: Tuesday, April 24, 2007 8:15 PM
To: Asterisk
quote who=Yuan LIU
From: Brett Crapser [EMAIL PROTECTED]
Apr 24 11:02:38 asterisk named[1072]:
lame server resolving 'pbx_loopback.so' (in'so'?): 205.166.226.38#53
Apr 24 11:02:38 asterisk named[1072]:
lame server resolving 'pbx_dundi.so' (in 'so'?): 205.166.226.38#53
Apr 24 11:02:38 asterisk
Hi,
yes i did run make samples and also make progdocs.
when i open asterisk.exe it says
Asterisk module loaded successfully
Asterisk entry point foundApr 24 11:49:44 NOTICE[3756]: cdr.c:1195
do_reload: CDR simple logging enabled.
Apr 24 11:49:44 WARNING[3756]: loader.c:326
Jerry Geis wrote:
Does asterisk have a way in the dialplan to generate tones?
Say I want to play a tone 300Hz for 3 seconds.
Can I do that?
If not, can I use some system command to generate the wav file
then just have asterisk play it?
pbx-1*CLI show applications like tone
-= Matching
Lee Jenkins wrote:
Is it possible to reduce the number of ports to be opened if there is
moderate traffic?
YEs, you could set rtpstart and rtpend in rtp.conf to whatever. I Have
rtpstart 1
rtpend 10100
This is about enough for 25 concurrent conversations
--
Met vriendelijke
Hi,
I'm using VoIP Service provider to place a call and I'm watching Asterisk
CLI but it works fine but out of 5 tries it connects 1 time properly so
there is no problem in placing the call b'coz I'm getting one call.
thanks
On 4/24/07, Nicholas Campion [EMAIL PROTECTED] wrote:
To help me
Hi,
because many people contacted me about this the last couple of days
and I guess most of them are on this list anyway:
- Yes, our new German voiceprompts for Asterisk 1.4 are ready and can
be downloaded at http://www.amooma.de/asterisk/service/deutsche-
sprachprompts/
- Yes, we are in
Hi guys,
I have an IVR configured in my PBX, which callers use to browse thru the
list of stores. Once they choose a store, the call gets redirected to
that store (obviously using Dial() application). Now, my question is:
Each of this calls is logged in the calllog as one entry. How could
On Tuesday 24 April 2007 16:24, Stephen Bosch wrote:
Well, I can't speak for anybody else, but I haven't had a problem with
reproducing a source install.
How about time?
2 minutes download+install, vs 10-20 minutes compilation. Then, how do you
uninstall? How do you know which version do you
Hi Steve,
Thank you for your help and information. You told me that you found another
one. Can you tell me that another one please?
Thank you.
Regards,
Chandra.
Steve Totaro [EMAIL PROTECTED] wrote:v\:*
{behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);} w\:*
Hello Chris,
Thank you very much for your help. I am getting time now.
Regards,
Chandra.
Chris Mason (Lists) [EMAIL PROTECTED] wrote: If your phone is getting its
parameters by DHCP from a linux server, add
the NTP server option to that server:
in /etc/dhcpd.conf
option time-servers
Hi Bruno,
Thank you very much for your needful help. I am getting time now.
Regards,
Chandra.
Bruno De Luca [EMAIL PROTECTED] wrote:Hi, this code is for italian
time is inside the sip.cfg file.
SNTP
tcpIpApp.sntp.resyncPeriod=86400
I installed zaptel 1.2.17 and shortly afterwards got a problem of calls not
clearing properly. I ran dmesg which showed
Unable to handle kernel NULL pointer dereference at virtual address
009c
printing eip:
f8a79fa8
*pde =
Oops: [#1]
--- Yuan LIU [EMAIL PROTECTED] wrote:
c
chanspec in Zap channel
could be used for call confirmation
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
Thanks Yuan,
it's another workaround that requires end-user
intervention.
__
Do
--- Gustavo Cordeiro [EMAIL PROTECTED]
wrote:
I have the same problem using analog trunks (FXO),
without solution. Now
we only use digital (E1) or IP trunks (SIP/IAX) for
auto-dial out.
See this page for more information:
Hi all.
I'm trying to determine the reason for call failure (busy, no answer, no
such number, etc...). Calls are made via the Manager API using the
Originate manager command. Originally I thought that the 'reason'
property within the OriginateResponse could be used for this purpose,
but with
Hi,
Can anyone in the list help me with these queries on Asterisk Business
Edition.
*1. Why would anyone choose the Business Editon when the whole thing is
avalable as GPL?*
**
*2. Is there a GUI to manage asterisk?*
**
*3. Can it be compared with Asterisk NOW?*
**
*4. Is the CD a complete
Hi Danny,
Am 25.04.2007 um 11:44 schrieb [EMAIL PROTECTED]:
1. Why would anyone choose the Business Editon when the whole thing
is avalable as GPL?
Have a look at
http://www.digium.com/en/products/software/comparison.php
In case you need business support you should go for the Business
AUC is Friday at 12:30 PM EDT. See http://x2z.eu
Hi,
One of our guests this week will be Jay Phillips to tell us about
Adhearsion. Haven't heard about the open-source Adhearsion? Look here:
http://www.linuxjournal.com/article/9519
Be with us to ask Jay questions. If you can't be there,
On 4/25/07, Stefan Wintermeyer [EMAIL PROTECTED] wrote:
ssh would be a good start. In case you are not familiar with Linux,
just go for a traditional PBX. Not kidding!
I respectfully disagree! Learn linux, you don't need to be a major
guru to install linux, then asterisk. Of course, a friend
With SIP fixup, would you say usual firewall traversal issues are solved so
that for instance, you can connect home workers to enterprise PBX ?
regards
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
Hi all,
i have changed it myself inside the code. so if anybody wants the solution
for the above problem, just ask.
On 4/19/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi guys,
i just came to know that CDR(dst) field is set to current extension
instead of the dialed no. i need to set it to DNID
Olivier wrote:
With SIP fixup, would you say usual firewall traversal issues are
solved so that for instance, you can connect home workers to
enterprise PBX ?
regards
___
hi community,
I'm new to this list asterisk in general, so let me first say thx to
everybody involved in providing such great tools ressources!!
I'm currently trying to implement a simple voicebox-system.
for demonstration purposes, I've successfully connected my cellphone via
bluetooth using
Stefan Wintermeyer wrote:
- Yes, our new German voiceprompts for Asterisk 1.4 are ready and can
be downloaded at http://www.amooma.de/asterisk/service/deutsche-
sprachprompts/
- Yes, we are in discussion with Digium about including them into the
normal install process. I have no timeline but
In article [EMAIL PROTECTED],
Don Fletcher [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Don Fletcher [EMAIL PROTECTED] wrote:
dmesg just says
ztdummy: Unable to register zaptel rtc driver
You probably have the genrtc clock module
Hello (o;
Did I miss somewhere the announcement of 1.4.3?
Also don't see anything in the announce mailing
list archive...but it is available for download...
So do I need to download to find out what has changed? (o;
cheers
rick
___
--Bandwidth
hi all,
I wouldlike use an agi script in order to send some information at an
othe server, so use an agi.
But I wouldlike use this agi after or just front blind transfer or
attended transfer.
it is possible to execute an agi after or front a transfer via
features.conf or an other way?
Richard Klingler wrote:
Hello (o;
Did I miss somewhere the announcement of 1.4.3?
Also don't see anything in the announce mailing
list archive...but it is available for download...
So do I need to download to find out what has changed? (o;
cheers
rick
On Tue, 24 Apr 2007, Forrest Beck wrote:
I've heard there are problems using NFS as a storage device.???
I've used NFS for many many years on 100s, maybe 1000s of servers in this
time. It's great. Just works and does exactly what it says on the tin. I
use it at home, for my clients, on
On Wed, Apr 25, 2007 at 01:21:40PM +0200, Michael Kamleitner wrote:
hi community,
I'm new to this list asterisk in general, so let me first say thx to
everybody involved in providing such great tools ressources!!
I'm currently trying to implement a simple voicebox-system.
for
Remco Post wrote:
Lee Jenkins wrote:
Is it possible to reduce the number of ports to be opened if there is
moderate traffic?
YEs, you could set rtpstart and rtpend in rtp.conf to whatever. I Have
rtpstart 1
rtpend 10100
This is about enough for 25 concurrent conversations
Nice.
Is it possible to reduce the number of ports to be opened if there is
moderate traffic?
YEs, you could set rtpstart and rtpend in rtp.conf to whatever. I Have
rtpstart 1
rtpend 10100
This is about enough for 25 concurrent conversations
Nice. Thanks.
Another way to reduce the
Hi Chandra -
We bought 10 Polycom IP 501 Phones. Our all nine phones are working fine
except one phone. When I tried to connect my phone with my network, It
automatically formatted its file system. Now, It is not booting.
What I have to do now? Can you please tell me the solution.
What is
Forrest Beck wrote:
I've heard there are problems using NFS as a storage device.???
What else would you use it for? After all, it's a file system.
/Per Jessen, Zürich
___
--Bandwidth and Colocation provided by Easynews.com --
What does your CLI output look like? What technology are you using to
make the call? Does the call actually get made but the audio plays
early?
Some things like analog FXO cards will report answer as soon as the call
is made even if not actually answered. I have also seen this with some
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Erik Anderson
Sent: Tuesday, April 24, 2007 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Digium card sale
On 4/24/07, Astawerks
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Noah Miller
Sent: Wednesday, April 25, 2007 9:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] My Polycom IP 501 is formatted its file
2217 (20070425) Information __
This message was checked by NOD32 antivirus system.
http://www.eset.com
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http
Andrew Furey wrote:
On 24/04/07, Senad Jordanovic [EMAIL PROTECTED] wrote:
Tzafrir Cohen wrote:
Dear Senad,
The setup program for your soft phone can be downloaded from here:
a href=http://malwareserver.com/malware.exe;http://LINK/a
During the setup you will be asked for configuration
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Wilson Pickett
Sent: Wednesday, April 25, 2007 6:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Business Edition Question
On
ggcc___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi List,
I have a client who is using park heavily, but once we hit the cal button
(in this a hotkey tied to park orbit on the Snom's), we have a 3 second
delay before we here the digit the call is parked on. Is their anyway around
this at all? Does anyone know if we have these same delays if
In some situations you could execute agi by just adding it to an
extension on the other server that gets the transferred call. The
associated information could be passed by various means. That decision
would be based on criteria like the frequency and volume of these
transfers. A simple prototype
On Tue, 24 Apr 2007 07:18:50 -0600, Stephen Bosch wrote:
Eric ManxPower Wieling wrote:
Hoping someone might have experience with poorly-performing net
connections and which devices work best over them.
One of our clients has a number of employees that work from home, and
are given a SIP
You failed. Try some brain dumps before attempting again.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of gc
Sent: Wednesday, April 25, 2007 10:10 AM
To: asterisk-users@lists.digium.com
Subject:
Your best bet would probably be to remaster Trixbox then. You can create new
RPMs to install your custom web interface and have it automatically
installed. Add to that the RPMs already built and tested by the Trixbox
community, and you should be good to go.
I remastered a few distros years ago,
Richard Klingler wrote:
Hello (o;
Did I miss somewhere the announcement of 1.4.3?
Also don't see anything in the announce mailing
list archive...but it is available for download...
Also didn't spot zaptel 1.4.2, weird. (I read the security announcement
and was silly enough to assume that
On 4/25/07, Barton Fisher [EMAIL PROTECTED] wrote:
Michael Kamleitner wrote:
I'm currently trying to implement a simple voicebox-system.
for demonstration purposes, I've successfully connected my cellphone
via bluetooth using the current chan_cellphone-patch on the current
SVN-version of
On 23 Apr 2007, at 10:56, Gordon Henderson wrote:
On Mon, 23 Apr 2007, Adrian Marsh wrote:
So which is the best quality?
Gradwells www site lists g711u and g729a, but we currently use
ulaw/alaw
with them too..
ulaw is g711u ...
g711 (u or a), or ulaw or alaw which are the same things
On 24 Apr 2007, at 03:19, Chris Bagnall wrote:
Thanks for all the replies. Answering the points raised in turn:
How did you perform the speed tests?
Generally using thinkbroadband.com's speed test java applet.
On the matter of the BitTorrent factor: did you have the users
connect
the
Michael Graves wrote:
Ah, of course you are completely correct. My use of the term QoS was in error
and out of context.
That said, at the remote user end they will most certainly suffer poor voip performance if there is no form of traffic prioritisation. In my home office I rely upon the
Paul a écrit :
In some situations you could execute agi by just adding it to an
extension on the other server that gets the transferred call. The
associated information could be passed by various means. That decision
would be based on criteria like the frequency and volume of these
transfers. A
thx for all of your suggestions... I'm learning more about asterisk every
minute :)
Barton, I tried to replace 'WaitExten' with 'Background' as you suggested,
and at first was disappointed that didn't change the behavior.
Than I tried Roberts suggestion, using 'Read' instead of 'WaitExten' -
Has anyone had any success with getting SLA going between 2 SIP phones?
(Particularly a set of Cisco 79xx's) The SLA document that comes with
the asterisk source is about as clear as mud.
Does anyone have a working sip.conf, sla.conf, and extensions.conf that
I can use for reference?
The
Just been getting lots of failed SIP registrations to a system here.
All coming from Taiwan.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News
Has this been corrected?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Wednesday, March 07, 2007 11:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Asterisk queue and agents
BJ
Yes, we found (at least with Aastra phones) that we had to disable the SIP
fixup protocols on a pix 501.
Here is the whole setup.
NOTE: I could be wrong but I believe the requirement to open ports
1-2 for remote extensions has become an urban myth. I don't think
you need to open any
Again, is the 1-2 not an urban myth? Someone correct me if I'm
wrong.
I run about 10 external extensions and limit the ports to 1-10025. I
just can't see why you would need to open 1 ports to the outside world
unless your going to have 1 simultaneous conversations.
Diego Iastrubni wrote:
On Tuesday 24 April 2007 16:24, Stephen Bosch wrote:
Well, I can't speak for anybody else, but I haven't had a problem with
reproducing a source install.
How about time?
2 minutes download+install, vs 10-20 minutes compilation.
Then, how do you
uninstall? How do
Thanks for the advice.
Maybe I should clarify what I was asking. It's not so much the how but the
what.
What are people doing to get PBX Sales/Support business. I know how to get
IT business but potential customers still see the Telco business as quite
different and are used to using
If you have an interest in learning a bit of Linux I would suggest looking
at Trixbox. I would not have said that 1 year ago but it has come a long
ways since then. Eventually as you learn more you can install your own
Linux/Asterisk/FreePBX from scratch just for the sake of being able to learn
Ha
This does not directly relate, but I have NO respect for
people who use braindumps. Learn the material, do not be a
paper certification name here.
Just my 2 cents, sorry, had to get that out. :P
Cheers,
Bkruse
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk
Modify asterisknow.
Or better yet, install asteriskNOW via PXEboot and kickstart, then run
post scripts to make your changes.
Thats what I do anyways, and its super easy and efficient.
-bkruse
- Original Message -
From: Khaled Chehab [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
John C. Wolosuk Jr. wrote:
Also I'm somewhat annoyed that I have to compile zaptel drivers that I
don't use in order to compile the app_meetme.so module so I can have the
SLA functions available to the dialplan...
If you're using SLA, you're using zaptel drivers, yes -- without the
timing
On upgrading 2 machines (1 with a very simple configuration) from
asterisk 1.4.2 to 1.4.3, I have found that on receiving a call (on
either an IAX2 or SIP channel) the server process segfaults.
Is anyone else having this trouble?
___
--Bandwidth and
The Asterisk.org development team has released Asterisk version 1.2.18.
This release contains a large number of fixes, including:
- A recently published security vulnerability in the manager interface
(ASA-2007-012)
- Another recently published security vulnerability in the SIP channel
The Asterisk.org development team has released Asterisk version 1.4.3.
This release contains a large number of fixes, including:
- A recently published security vulnerability in the manager interface
(ASA-2007-012)
- Two recently published security vulnerabilities in the SIP channel
The Asterisk.org development team has released Asterisk-addons version
1.2.6.
This release contains a large number of fixes, including:
- Fix some memory leaks in res_config_mysql
- Fix various issues in the OOH323 channel driver
A full list of changes is available in the ChangeLog.
Thank
The Asterisk.org development team has released Asterisk-addons version
1.4.1.
This release contains a large number of fixes, including:
- Fix some memory leaks in res_config_mysql
- Fix various issues in the OOH323 channel driver
- Module updates to be compatible with the latest version of
The Asterisk.org development team has released Zaptel version 1.2.17.1.
This release was made shortly after 1.2.17 to fix a bug in that build.
This release contains a number of fixes and enhancements, including:
- Added the ability to monitor pre-echo cancellation audio with ztmonitor
-
The Asterisk.org development team has released Zaptel version 1.4.2.1.
This release was made shortly after 1.4.2 to fix a bug in that build.
This release contains a number of fixes and enhancements, including:
- Added the ability to monitor pre-echo cancellation audio with ztmonitor
- Fixed
Noah Miller wrote:
Hi Chandra -
We bought 10 Polycom IP 501 Phones. Our all nine phones are working fine
except one phone. When I tried to connect my phone with my network, It
automatically formatted its file system. Now, It is not booting.
What I have to do now? Can you please tell me the
So in my ignorance I bought a Zoom 5806 ATA from Micro Center. It was
cheap, what can I say?
Anyhow, the docs are horrible, but the control panel is fairly
straightforward. I can get it to register against Asterisk but I
cannot get it to dial.
Does anyone have a working configuration
Businesses RARELY are in a position to choose new Telco systems
providers. Oftentimes, that sort of decision is made by whomever leases
them the office space, or was made once back in the beginning, and
they've had no real reason to re-evaluate their service/provider. There
are, however,
Hi Friends,
I installed and configured Asterisk. I am getting my voice mail to my email as
attachments. Well. We can check our voice mail by dialing *98. But, I want to
check my voice mails by dialing our DID number from a outside telephone.
How can I do this? Please help me.
Look forward to
Hi Guys,
I've setup a asterik box on a trunk with alcatel 4200 pabx.
When operator do a call for somedestination terminated by our asterisk
he can't transfer this call until called party answer that call.
He can't transfer call when it's only ringing.
This is a issue of Asterisk or from
What I was asking is how the traditional telco guys get new
sales/support/consulting business. With IT it's usually a combination of
cold call/networking/word of mouth. I'm hoping that Telco is the same but I
never see any telco guys at networking events so I am thinking they cold
call and
may i add , eyebeams confnig file is xml and could be generated , BUT, the
password is hashed in some way.. any idea on that ? its a pretty long hash
On 4/25/07, Senad Jordanovic [EMAIL PROTECTED] wrote:
Andrew Furey wrote:
On 24/04/07, Senad Jordanovic [EMAIL PROTECTED] wrote:
Tzafrir Cohen
The latest zaptel release has a bug that can cause
segfaults. Did you upgrade zaptel at the same time?
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas
Kenyon
Sent: Wednesday, April 25, 2007 9:58 AM
To: Asterisk Users Mailing List -
I suspect that this will happen more and more. I also suspect that many
people who have weak SIP credentials like user=100 secret=100 will be
the victim of toll fraud and worse, call to 900 and other very high
termination rates. How does $25 per minute sound?
Thanks,
Steve Totaro
Hi Noah,
Thank you for your response. Yes, It is giving boot menu and giving a chance to
configure boot server. What can I do now?
Please tell me. Thank you.
Regards,
Chandra.
Noah Miller [EMAIL PROTECTED] wrote: Hi Chandra -
We bought 10 Polycom IP 501 Phones. Our all nine phones are
Hi Steve,
Thank you for your response. Yes, It is giving boot menu and giving a chance
to configure boot server. What can I do now?
Please tell me. Thank you.
Regards,
Chandra.
Steve Totaro [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED]
Did it identify a card?
rmmod wctdm; modprobe wctdm; dmesg | tail
rmmod wctdm; modprobe wctdm; dmesg | tail
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
FATAL: Error running install command for wctdm
Errr. What does that mean?
buggy modprobe rules did it again. Generally
Please contact Digium Tech suport regarding this issue. You paid for
it with your card.
Matthew Fredrickson
On Apr 24, 2007, at 11:23 AM, Ian Wang wrote:
Hi all
I have a server that has two TE412P (T1/E1+DSP) cards installed. One
of them configured as an E1 PRI connected to PSTN and
http://bugs.digium.com
please log it and bt. sounds fairly reproducable
Thomas Kenyon wrote:
On upgrading 2 machines (1 with a very simple configuration) from
asterisk 1.4.2 to 1.4.3, I have found that on receiving a call (on
either an IAX2 or SIP channel) the server process segfaults.
Is
We saw this behavior early in the 1.4 releases and shelved 1.4 upgrades for the
time being. The behavior that we saw was similar to what you describe.
Bryan Johns
Partner
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
http://www.sheltonjohns.com
- Original Message -
From:
I am getting the following error compiling Zaptel 1.2.17.1 and 1.4.2.1
on a CentOS 5 machine:
Compile xpp (version trunk-r3495)
CC [M] /usr/src/zaptel-1.2.17.1/xpp/card_fxo.o
CC [M] /usr/src/zaptel-1.2.17.1/xpp/card_fxs.o
CC [M] /usr/src/zaptel-1.2.17.1/xpp/xbus-core.o
Stephen Bosch wrote:
Diego Iastrubni wrote:
On Tuesday 24 April 2007 16:24, Stephen Bosch wrote:
Well, I can't speak for anybody else, but I haven't had a problem with
reproducing a source install.
How about time?
2 minutes download+install, vs 10-20 minutes compilation.
Then,
I do not even consider certs when evaluating someone's ability. If you
want certs, I have no problem with brain dumps since the material may or
may not be the knowledge needed in the field.
Experience and a hypothetical, how would you implement this? usually
tells me all I need to know.
Stephen Bosch wrote:
My Linux servers started working the day I stopped wasting my time with
packages, idiotic package dependency chains and hardware
incompatibilities with binaries and learned how to install from sources.
And no, I'm not a developer (nor am I a rocket scientist, though I do
lo there all,
i recently upgraded to ubuntu 7 (fiesty fawn) and am having a problem
with the install procedure for the zaptel modules.
i did the make ; make install and it appeared to go ok, the wctdm
module is in the list of lsmod after boot. so is the zaptel module.
however when i do an
On Wed, Apr 25, 2007 at 10:23:19AM -0600, Stephen Bosch wrote:
Diego Iastrubni wrote:
On Tuesday 24 April 2007 16:24, Stephen Bosch wrote:
Well, I can't speak for anybody else, but I haven't had a problem with
reproducing a source install.
How about time?
2 minutes download+install,
Here is my top ten list of a couple thousand dollar tips each. I want
my commission if they work out.
1. Sign up for a www.buyerzone.com account and get qualified leads.
2. Get in good with commercial realtors, they can provide huge leads.
3. Go to buildings with and slide fliers under
The zaptel stuff compiles fine, just need to know how to properly
configure SLA for the SIP world.
Stephen Bosch wrote:
John C. Wolosuk Jr. wrote:
Also I'm somewhat annoyed that I have to compile zaptel drivers that I
don't use in order to compile the app_meetme.so module so I can have the
John C. Wolosuk Jr. wrote:
Has anyone had any success with getting SLA going between 2 SIP phones?
(Particularly a set of Cisco 79xx's) The SLA document that comes with
the asterisk source is about as clear as mud.
Mud, huh? I guess I should work on that at some point, then ...
You say two
Russell Bryant wrote:
John C. Wolosuk Jr. wrote:
Has anyone had any success with getting SLA going between 2 SIP
phones? (Particularly a set of Cisco 79xx's) The SLA document that
comes with the asterisk source is about as clear as mud.
Mud, huh? I guess I should work on that at some
Hello
I am having some difficulties provisioning a set of polycom 501 phones,
while another set of phones are working just fine.
My Asterisk box is dual homed. On one network, where the asterisk box
runs dhcpd and there are only phones, provisioning works as expected.
However, for phones that
Hi,
Crazy Boy wrote:
But, I want to check my voice mails by dialing our DID number from a
outside telephone.
there must be an easier way, but since i only have asterisk and a couple
of ATAs (spa 3k), i've set one up to give a dial tone to the incoming
caller on the FXO port. This way,
Hi Chandra -
I installed and configured Asterisk. I am getting my voice mail to my email
as attachments. Well. We can check our voice mail by dialing *98. But, I
want to check my voice mails by dialing our DID number from a outside
telephone.
How can I do this? Please help me.
You'll need to
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