Hi All,
Sorry if this has been around a millions times.. I have been off this
list for a few months now..
I have installed the latest asterisknow (upgraded asterisk to 1.6 as
well) and I am having a hard time getting my X100P and TDM400P working..
Its all new to me with dahdi because my old
WipeOut wrote:
Hi All,
Sorry if this has been around a millions times.. I have been off this
list for a few months now..
I have installed the latest asterisknow (upgraded asterisk to 1.6 as
well) and I am having a hard time getting my X100P and TDM400P working..
Its all new to me
Hi,
My Asterisk box has sat happily doing its job for years now and never
had any real issues.. Unfortunately a lightning storm the other night
appears to have damaged the TDM400P (1 FXS and 1 FXO port)..
Since this system was put together there seem to heave been a lot of
developments in the
Anyone had any experience with an Asterisk server as a VMWare virtual
machine?
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Mike Clark wrote:
Michiel van Baak wrote:
On 18:51, Tue 23 Oct 07, WipeOut wrote:
Anyone had any experience with an Asterisk server as a VMWare virtual
machine?
We are running multiple sites as a VMWare virtual machine.
All of them are voip only, so I have no idea how it works
Hi,
Ok.. I know dual NAT is a problem for SIP..
ie. UA - NAT - Internet - NAT - Asterisk
What about Multi-NAT where a dedicated public IP is mapped to the
private IP of the asterisk box..
ie UA - NAT - Internet - Multi-NAT - Asterisk
http://www.draytek.co.uk/support/kb_vigor_multinat.html
Thanks Steve.. So its the same as the dual NAT scenario.. :)
Steve Totaro wrote:
I have tried it with the best result of one way audio after spending a
few days doing everything imaginable. This is the only scenario where I
suggest using IAX.
Thanks,
Steve Totaro
WipeOut wrote:
Hi
WipeOut wrote:
Hi,
Here is the situation.. My Dad is working on contract in overseas.. He
has internet access in his hotel.. He wants to be able to talk to my Mum
but the calls are expensive..
I have an asterisk box setup for my business and it has a public IP
etc.. My Mum has access
Time Bandit wrote:
Can anyone post a sample of whats needed in iax.conf for an IAX UA to be
able to make and receive calls?
[7011]
type=friend
secret=S0m3S3cur3P4ssw0rd
qualify=no
notransfer=yes
[EMAIL PROTECTED]
host=dynamic
disallow=all
allow=ulaw,alaw,gsm
context=from-internal
Hi,
Here is the situation.. My Dad is working on contract in overseas.. He
has internet access in his hotel.. He wants to be able to talk to my Mum
but the calls are expensive..
I have an asterisk box setup for my business and it has a public IP
etc.. My Mum has access to a working phone
hi,
Is it easy to add to the AsteriskNow system?
I am looking to use it to replace my older Asterisk box but I put my
asterisk box on the internet and restrict access to specific IP
addresses with APF firewall.. So it would be nice if I could install APF
on AsteriskNow..
Thanks..
I am sure its been discussed before but I couldn't find it in my searches..
Looking to replace my Asterisk box (Ver 1.0 still I think) and really
like the idea of an easy to use gui to manage it.. I see the contenders
appear to be Asterisknow and Trixbox..
Has anyone player with both who can
Has anyone had any experience with this router??
I am looking to use it because I want to use a DECT phone in conjunction
with VoIP and this seems to check all the boxes for Wi-Fi, ADSL and VoIP
all at a good price.. I have never used Speedtouch hardware before so
any feedback would be
Hi,
I am investigating getting a wifi VoIP phone because its may be a better
option than an ATA and a cordless phone..
Does anyone have any experience with the whats out there??
Do they support things like WPA etc??
I have heard the battery life can be a problem.. Is this the case?
James Harper wrote:
I was looking for something like this a while back (actually, a wifi +
gsm combo), and came to the conclusion that a dect + gsm phone would be
a better option, except that they don't exist (much).
Maybe a VoIP capable DECT base station would be a better option for you?
These
Thanks Tom and Justin..
I did think they were separate entities.. I will pull down [EMAIL PROTECTED]
and see where I get to from there..
Justin Biggs wrote:
Another FYI: The latest [EMAIL PROTECTED] release (2.8) includes FreePBX
(sounded like you thought they were seperate entities). It
I see that [EMAIL PROTECTED] and FreePBX are going along similar lines with
web based management interfaces..
My Asterisk box has one analog phone, one analog line, 3 SIP phones, IAX
inbound numbers, an IAX outbound trunk, IVR menus and voicemail boxes in
different contexts for each of the
Hi..
I have to setup an extension in a remote location that will use a
cordless analog telephone.. I am looking at the IAXY to do this for
me..Basically the data path will be as follows...
[Asterisk] == (NAT) == {Internet} == (NAT) == ATA -- Handset
Since there are two NAT boxes in the path
Kevin P. Fleming wrote:
WipeOut wrote:
To save bandwidth I would like to stay away from using the G.711
codecs.. Does the IAXY support GSM or iLBC? I couldn't find anything in
the docs..
No. The IAXy only supports G.711 ulaw/alaw and ADPCM.
I don't know what 'docs' you were looking
Hi,
I am looking for a budget IP phone that can use preferably iLBC or GSM
codecs..
Suggestions?
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Hi,
I'm just trying to setup a Grandstream BT-102 that I had lying around
and haven't used for over a year..
I loaded up the 1.0.6.7 firmware and factory reset the config..
I have it working but there is no sound to the handset ear piece.. On
speaker phone it works fine in an echo test..
Hi,
Just pulled out the BT-102 because I need to use it again, entered in
the TFTP server to get the latest firmware so its now in 1.0.6.7 and i
now was to factory default the phone and set it up from scratch..
I tried the instructions (copied below this message) from the latest
available
Hi,
Thinking about an IVR application and trying to get a handle on the best
way to structure it so that the maximum number of concurrent calls can
be achieved..
If the voice prompts were stored in a GSM format and were being played
out through an IAX trunk that uses GSM compression would
Dave Brooks wrote:
Hi.
I'm new here so I hope this is a sensible question/sensible place for it.
I have a PSTN to IAX phone number with voipuser.org that I'm using to
test an IVR service. The only trouble is that after approximately 40
seconds of silence (e.g. after background playback of a menu
Michael Vogel wrote:
Hi!
Some - few - providers are using IAX2 as a protocol. Most are using
SIP. I know that there are advantages of IAX2 regarding multiple
connections. But beside this I'm asking myself (and you all) why I
should prefer IAX2 when my SIP connection is working.
Are there
Craig Waddington wrote:
Have a strange problem.
2 different asterisk servers, running different CVS.
One behind Firewall, one not.
Cisco 7940 phones.
Over the past two weeks, users have had a problem with one way audio,
after about 2 minutes into a call, they can hear the other person, but
the
Craig Waddington wrote:
I found this:
http://lists.digium.com/pipermail/asterisk-dev/2003-May/000764.html
But it is old, and I am sure lots of changes have been made to the
source, since then.
Where and how do you set absolutetimeout=0, would this help?
A test I want to perform is, we make a call,
Please can someone look at my last two posts and try and shed some light
onto why my system is dropping calls..
If I don't get it right we will be forced to drop Asterisk which I
really don't want to do..
Thanks..
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Jason Williams wrote:
On Tue, 23 Nov 2004 13:17:57 +, WipeOut
[EMAIL PROTECTED] wrote:
Please can someone look at my last two posts and try and shed some light
onto why my system is dropping calls..
If I don't get it right we will be forced to drop Asterisk which I
really don't want to do
www.bicomsystems.com
USA 1-212-400-7921
UK 0870 682 782
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: 23 November 2004 14:18
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] NEED HELP
Senad wrote:
Hi Senad,
No, this server that is having the issue is behind a NAT firewall
connecting through IAX to a termination provider.. Being IAX I would
think NAT is irrelevant.. Also there is no patten to the dropping of
calls, a call can last less than 1 min or over 30 min or anything in
Geoff Nordli wrote:
onto why my system is dropping calls..
If I don't get it right we will be forced to drop Asterisk which I
really don't want to do..
Thanks..
Did you know that you can obtain commercial support for Asterisk?
http://www.digium.com/index.php?menu=software_support
I am
Eric Wieling wrote:
WipeOut wrote:
The problem is that I don't think Asterisk is causing the problem
(not entirely anyway), I think its the internet and that IAX is too
sensitive to packet loss so when the packet loss exceeds a certain
threshold it just drops the call instead of trying
Hi,
Didn't get any opinions on the log file I mailed onto the list over the
weekend so I am continuing to try and track the cause for the dropped
calls..
I have a feeling that its to do with IAX being way too sensitive when it
comes to packet loss.. Since it is going across the internet it
Mike Dent wrote:
Hi,
Anybody here from the UK using Asterisk at home?
I'm looking for a SIP phone which will work with Asterisk and
not leave me broke!
I got one of the Tecom ones from Solwise but it refuses to
login to Asterisk server for some reason. May have to send it back.
What are the other
Hi,
I need help finding why my system is dropping calls..
I enabled debugging on my box in the hope it would lead me to the answer
as to why my system is dropping calls but unfortunately nothing is
jumping out at me..
I have attached the portion of the messages file for two calls that were
I have just upgraded from Asterisk 0.7.2 to 1.0.2 and seem to be having
an issue with the zaptel init script..
If I run..
#modprobe zaptel
#modprobe wcfxo
#modprobe wcfxs
.. from a command line it load and appears to be working fine..
If I try and use the init script I get errors about
for that, it has solved the problem..
-- Original message --
From: WipeOut [EMAIL PROTECTED]
I have just upgraded from Asterisk 0.7.2 to 1.0.2 and seem to be having
an issue with the zaptel init script..
If I run..
#modprobe zaptel
#modprobe wcfxo
#modprobe wcfxs
.. from
John Millican wrote:
-- Original message --
From: WipeOut [EMAIL PROTECTED]
I have just upgraded from Asterisk 0.7.2 to 1.0.2 and seem to be having
an issue with the zaptel init script..
If I run..
#modprobe zaptel
#modprobe wcfxo
#modprobe wcfxs
.. from
9. When the hardware checker finds a new Tiger Jet device, just
ignore it. (Anyone know how to make it stop bothering me?)
Choose Do nothing and it should stop bothering you.. thanks for the
install tops I may be having a go with FC3 in the near future..
Hi,
Is there any method to log the reason a call was ended / terminated /
dropped??
I am getting a fairly high nimber of calls being dropped but have no way
of working out why.. I need to still upgrade Asterisk to ver 1.0 but I
still need a way to track the reason for the call dropping so that
and has to reconnect quite often..
Anyone got any other ideas to try and stop it messing up my internet
connection cos its causing havoc with my VoIP calls coming in and going
out over the ADSL line..
Later..
WipeOut wrote:
Hi,
This may be one for the broadband guru's out there..
I have a single
Hi,
This may be one for the broadband guru's out there..
I have a single analog line coming into the house.. This line is for my
ADSL and home phone.. My Asterisk box uses an X100P card to connect to
the analog line.. I have a microfilter on the line etc.. The rest of my
phone system works
Stewart Nelson wrote:
I have a single analog line coming into the house.. This line is for
my ADSL and home phone.. My Asterisk box uses an X100P card to
connect to the analog line.. I have a microfilter on the line etc..
The rest of my phone system works inbound and outbound calls via a
VoIP
I always just let the phone poll the Snom update server for updates but
while the server is back at version 2.03o the latest stable downloadable
version on the website is 2.04n..
Is Snom not distributing updates for the 200 from their server anymore??
Bastian Schern wrote:
WipeOut schrieb:
I always just let the phone poll the Snom update server for updates
but while the server is back at version 2.03o the latest stable
downloadable version on the website is 2.04n..
Is Snom not distributing updates for the 200 from their server anymore
hi,
I am looking to upgrade the firmware on my GS phone but the site doesn't
have the IP adress of the TFTP server anymore or anywhere to download
the firmware..
Does anyone know this information?
What is the current stable firmware version?
Later..
Duane wrote:
WipeOut wrote:
hi,
I am looking to upgrade the firmware on my GS phone but the site
doesn't have the IP adress of the TFTP server anymore or anywhere to
download the firmware..
Does anyone know this information?
Can get it off the web: http://hellofone.com/downloads/
What
Trilogy India wrote:
Hi,
I want to know, if someone has tried to use clustering
in asterisk to increase its scalability and make it
distributed??
If yes, how easy it is to cluster?
Can someone please ive me details about the same
Thanks
Varun
This has been discussed a number of times in the
Dorian Gray wrote:
Kevin Walsh wrote:
Leif Madsen [EMAIL PROTECTED] wrote:
Also, from what I have been told (and I've tested this by building
zaptel, but not any of the other sources) is that you no longer need
the sourcecode with the 2.6 kernel. You can create a symlink to:
/lib/modules/`uname
Hi,
Would be interestd in anyones ideas for this problem..
We are starting a new division to our company, the people in this new
division will be the same people who are on the old division..
Calls for each division come in on seperate numbers and go through
seperate menus but ring to common
Senad Jordanovic wrote:
Can anyone think of any easier ways?
How about if you put second division on different server, and then share
VM storage on the network between two asterisk boxes?
SJ
The single server works fine for the two divisions making and recieving
calls..
Its that each
Jason Williams wrote:
At 11:40 29/06/2004 +0100, you wrote:
Senad Jordanovic wrote:
Can anyone think of any easier ways?
How about if you put second division on different server, and then
share
VM storage on the network between two asterisk boxes?
SJ
The single server works fine for the two
Hans-Henrik Andresen wrote:
Hi,
A'm about to set up a asterisk for 5000 users, and the customer had a 64bit
server - can asterisk compile on that ? I will use a digium X100P for timing
use will that do on a 64bit ? (I'm using SUSE91 kernel 2.6)
What else ? Is it posible to have only one server for
Setting up a new system using Fedora Core 2..
Tried following the instruction below (from the mailing list archives)
that worked before..
cp configs/config-for-my-kernel .config
make oldconfig
make include/asm
make include/linux/version.h
make SUBDIRS=scripts
.. but now the FC2 kernel has been
Hi All,
I decided to have a go at installing Asterisk on FC2 which now runs on
Kernel 2.6..
Unfortunately I didn't get very far..
When trying to build zaptel it required me to link /usr/scr/linux-2.6 to
the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess
thats still the RH
Joshua M. Thompson wrote:
On Thu, 2004-05-20 at 05:12, WipeOut wrote:
When trying to build zaptel it required me to link /usr/scr/linux-2.6 to
the default source dir which is /usr/src/linux-2.6.5-1.358.. I guess
thats still the RH infulence.. :)
After than I tried again but the page rolls
Joshua M. Thompson wrote:
On Thu, 2004-05-20 at 12:04, WipeOut wrote:
Thanks for the try but its didn't work.. Got exactly the same result..
Apparently the FC2 2.6.5 kernel has another issue, one that I didn't
start seeing until 2.6.6 (I build my own kernel RPMs.) There are a few
files
Hermann Wecke wrote:
Sorry to ask this here but I believe that it is the best place to receive
a feedback...
I would like to know if anyone is using SNOM 100 / SNOM 200 phones with *,
and the overall impression about these phones...
I am using Snom 200's and they work great.. I would guess the
No this is not another of the What hardware do I need? posts.. :)
Just wondering if anyone has calculated the memory consumprion for
running asterisk..
For example, when its idle it uses U MB or RAM, uses V MB for each
active Zap channel, W MB for each active SIP channel, X MB for each
into faster processors to handle more calls and services..
Thats really all I was talking about so that it helps people size their
systems and would probably mean fewer What system do I need? questions..
Later..
WipeOut wrote:
No this is not another of the What hardware do I need? posts.. :)
Just
Steve Kennedy wrote:
On Wed, Apr 21, 2004 at 03:24:06PM +0100, Craig Waddington wrote:
Currently using voiptalk.org and the quality is getting really bad.
I would like a second provider preferably in UK, anyone got any suggestions?
That's the trouble with running VoIP over contended
Darren Nickerson wrote:
Folks,
I recently swapped a TDM400 FXS card that was working perfectly into a new
server (running recent CVS), and it's either misbehaving (unlikely), or I've
missed something obvious (much more probable). Everything seems to be
working, but I can't get any dialtone from
Hi,
I have a Snom 200 that has had admin mode switched off and I have no
idea when the admin password has been set to.. Does anyone know of a way
to reset the phone to factory defaults??
Later..
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[EMAIL PROTECTED]
a blank password or
it did not set it to be blank..
I will have to get hold of the distributor next week..
Later..
On Sat, 17 Apr 2004, WipeOut wrote:
Hi,
I have a Snom 200 that has had admin mode switched off and I have no
idea when the admin password has been set to.. Does anyone know of a way
imagine you might possibly
have to pay them an unlock charge so you can change carriers.
Or did you accidently set the admin password?
Chris
On Sat, 17 Apr 2004, WipeOut wrote:
Hi,
I have a Snom 200 that has had admin mode switched off and I have no
idea when the admin password has been set
Pertti Pikkarainen wrote:
There is a way.
Right after reboot, and when you see the first text, hit any key
and you will get a 'boot menu'. Give the phone an ip-address and
define a tftp-server.
The bootfile must be named snom200.bin ( e.g rename the latest snom sw ).
After you have
Olle E. Johansson wrote:
We're getting closer and closer to a 1.0 release of Asterisk. In order
to get there,
the development is now 110% focused on solving major, critical and
crash bugs.
(And yes, if you follow the CVS updates, you'll see the impossible
extra 10% :-)
All sounds very
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
WipeOut [EMAIL PROTECTED] wrote:
FC1 is basically what RHL10 would have been so compatibility is really
the same as for RH9, the only issie is there appears to be an issue with
the version of bison than comes with FC1 and Asterisk
Paul Tyreman wrote:
Thanks for all the replies.
Can someone tell me if it is possible to put two of these X100P cards
into the same machine to order to gain access to two BT landlines ?
Would it also be possible for someone to outline in a bit more detail
the procdue for limiting which
Paul Tyreman wrote:
What I want to do is have the asterisk server sat in my house and used
by my family to access the BT landline and to recieve calls made to
that landline. If it is not possible to do the auto attendant thing
then so be it, I will just have all phones in my house ring when a
Victor Perez wrote:
Has anybody tried to install * in any of these minimalist linux distros like tinylinux?
Which linux distro would you use to run * in old P2, P3 boxes?
I have got it to install on Trustix (92MB min install) but I have moved
to Fedora now for other reasons..
Brancaleoni Matteo wrote:
I made a custom fedora mini distro, something like
350 megs, including apache,php,mysql webmin
of course installable from a cd in 20 minutes, more or less :)
at the end you have a fully working asterisk installations,
along with some basic tools like webmin and
a full
Jain, Sonal wrote:
Is it true that every time we make a change in the configuration file we need to restart the asterisk server. This will not be practical in the production environment.
Thanks,
No, you don't have to restart, you have to reload..
From the CLI just type reload and hit enter..
[EMAIL PROTECTED] wrote:
Hi there
Does anyone know if it is possible to install a largescale asterisk
cluster with up to 1000 external lines.
Redundancy and loadbalancing would surely be a must for a such system,
but which other things should be considered?
Are you planning on using analog or
James Gardiner wrote:
Hi *ers,
I recently got an Email from Redhat about the dropping of support for Redhat
9 on the 30 of April and that Fedora Project is the recommended future,
otherwise, RedHat enterprise ($$$).
Yup, this has been coming up for a while now..
Considering this, I would like
Adam Hart wrote:
WipeOut wrote:
Doesn't NuFone use SER in front of Asterisk? so using asterisk purely
as the PSTN gateway..
Later
Nufone offers IAX termination, SER is SIP - or am I missing something
here?
Sorry, I was not thinking, you are correct..
Just most termination providers (SIP
Martin wrote:
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686
-DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-04/03/04-10:19:04\
-DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\
-DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\
-DASTVARRUNDIR=\/var/run\
For anyone who is interested..
I downgraded my system to..
asterisk-0.7.2
libpri-0.5.2
zaptel-0.8.1
+asterisk-addons
..and its all working again...
Later..
WipeOut wrote:
Hi,
I have just built my home Asterisk box into a better PC that became
available (still only a P2 350 but it only has
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
Hi,
I have just built my home Asterisk box into a better PC that became
available (still only a P2 350 but it only has to manage 1 analog line)..
Anyway I have built it on Fedora Core 1.. I have an X100P and a TDM400P
(1 module installed)..
These cards were working fine in my older PC that
Angus Berry wrote:
I haven't found this in any docs or faqs yet, so I'm wondering if I can
achieve what I would like to do.
On an Asterisk PBX with multiple PSTN lines, I'd like to call in from
one PSTN line, probably via cellphone and access the PBX as if I were
local to it. From here I'd like
Looks like a cool system.. looking forward to seeing it develop..
Later..
Areski wrote:
Hello Asteriskos,
Screenshot:
http://www.areski.net/asterisk-meetme/about.php
The goals of this application is to control your audience/users in the
conference room. That will allow you to have a visual
Matt Bridges wrote:
I've configured my [*] to dial the pstn which is working like a charm.
I've also configured an extension to ring when the PSTN line is ringing
which is also working brilliantly, but, sometimes it doesn't detect that the
call has been hungup.
I've had a look on voip-info and
Hi,
Having a small problem here and wondering if anyone else has seen it..
My Asterisk box is behind NAT so I need to register with the external
IAX Asterisk boxes for calls to be received..
Up till yesterday I only needed to register with a single external IAX
server and all was working
[EMAIL PROTECTED] wrote:
Works like a charm for me.
I have both VoicePulse and NuPhone registered in IAX.
Depending upon the phone nr dialed, I send a call via NP or
VP.
And yes, my [*] box is behind a NAT.
Include the relevant lines of your iax.conf so we can take a
look.
Cheers, Willy
There is
Rich Adamson wrote:
Gus,
There is nothing to it really register lines are pretty simple..
register = user1:[EMAIL PROTECTED]
register = user2:[EMAIL PROTECTED]
From the cli iax2 show registry only shows the first entry..
These are for inbound services not outbound, I didn't think it was
Rich Adamson wrote:
Thats the problem I have a dynamic IP on my side which is why I need the
register line in the iax.conf..
iax2 debug shows that it is registering the first line but not the second..
I assume you've tried the easy stuff... separate the two statements with
some additional
!
In the iax.conf file I have broken the inbound and outbound into 2 separate
stanzas, i.e. -
; for inbound from Nufone
[NuFone]
Type=user
; for outbound to Nufone
[NuFone-peer]
Type=peer
Hope this helps.
Ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
simprix wrote:
What kind of specs do I need for a asterisk box that will have a pri for
pstn and about 65 sip phones
I was thinking a Xeon 3.05
What length is a piece of string when you cut it?
I was thinking 2.374 m
Sorry about the sarcastic answer but if you look through the mailing
list
Peer Oliver schmidt wrote:
Brian Capouch wrote:
It might be helpful for us all if the author could let us know more
about the environment in which this application was built. .
I'm getting all kinds of errors when I try to run it, and I suspect
that either my Postgres or PHP installations are
Peer Oliver schmidt wrote:
Wipeout wrote:
Another thing I had to do was changing the defines.php file to
reflect my environment. After that, things went smooth.
On my server the links dont even work in the menu on the left.. Not
sure what is going on with the code and dont have the time
Tony Mountifield wrote:
I posted this a week or two ago but no replies, so trying again...
Summary: Two phones in different locations, each behind NAT, can both
talk to an Asterisk server on the net, for the demo or for voicemail,
but can't maintain a call to each other via that asterisk.
Carlos Chavez wrote:
I have been trying out Asterisk with the speex codec with X-lite as a
client. I applied the REG patch on my windows machine that is recommended in
Voip-info.org. Every time I make a call I get the following error:
codec_speex.c:167 speextolin_framein: Out of buffer
Daniel Bichara wrote:
WipeOut wrote:
Carlos Chavez wrote:
I have been trying out Asterisk with the speex codec with X-lite
as a
client. I applied the REG patch on my windows machine that is
recommended in
Voip-info.org. Every time I make a call I get the following error
Brian Capouch wrote:
Olle E. Johansson wrote:
Do *not* send out personal replies on the list.
Yes! Yes!! Yes!!!
Let's change the way the list software works so people won't get
hammered by replying and rid this list of that pox once and for all.
B.
The biggest problem with having replies
Alessio Focardi wrote:
Quick hint:
do I need cgi or cli version of php to interact with asterisk agi ?
I'm using cgi now, with strange results
tnx !
You need the CLI binary for PHP to work with AGI
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Asterisk-Users mailing list
Umar Sear wrote:
Hi I really hope somebody can help me out.
I have an asterisk installation working on a Redhat 9 system. I now
want to add the MySQL functionally to it. However when I make the
necessary changes, (downloading the add-ons, and changing the Make
file) the make fails.
I
I thought thats what http://www.iaxtel.org was all about.. :)
Daniel Bichara wrote:
Hi All,
We know everyone can offer services. May we build a interconnected *
network all over the world to offer best conditions each other? We can
set a service level agreement and try ;-)
Any one?
Joao Carlos Moura wrote:
I need to develop an web interface to include clients automatically in
Asterisk. So, to make this possible I need
that all my peers and exten being at a database (Mysql).
Where do I find doc´s regarded for it?
Thank you very much,
J Moura
I think MySQL friends is
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