Steve, Chris :
I too had this problem and the solution was not tweaking
the AMD parameters, but playing a short audio file (even
a really really short one) before executing the AMD function.
The key is executing the Background step before AMD()
Please see sample dialplan below :
On Nov 23, 2007 11:10 AM, Vincent wrote:
On Wed, 21 Nov 2007 15:45:35 -0500, Baji Panchumarti
STAT() and record() are doing exactly what they are
supposed to. Use the s flag to fetch the file size. You
have to try a few hangups and figure out a minimum
file size that qualifies
Don't know if they are related, look for 26 on this page:
http://www.freepbx.org/support/documentation/howtos/howto-resolve-freepbx-and-sipura-linksys-feature-code-conflicts
--
On Nov 21, 2007 10:45 AM, Jarga Jallow wrote:
I have a Linksys sipura phone which does not dial ext 26 only,
On Nov 21, 2007 2:51 PM, Vincent wrote:
Thanks for the tip, but it doesn't seem to work:
==
[...]
==
Looks like Record() always creates the file, even
if the user hung up without leaving a message.
Any other idea?
STAT() and record() are doing exactly what they are
On Nov 20, 2007 6:24 AM, Eric ManxPower Wieling wrote:
[...] but do generally have an e-mail-SMS gateway.
Check with your carrier.
http://en.wikipedia.org/wiki/SMS_gateways
--
___
--Bandwidth and Colocation Provided by
page 511
use dialplan function STAT()
--
On Nov 20, 2007 9:42 PM, Vincent wrote:
Hello
I didn't find the answer in the ATOF 2nd Ed: When using the Record()
application, I need to know how it ended: Did the user leave a
message, or did he hang up?
If the latter, Asterisk stops right
On Nov 19, 2007 3:45 AM, Abdul wrote:
Hi all,
I was playing with asterisk .gsm sound file to work for callback.
But the quality is very poor and sound is very low so we cannot clearly hear
what is sound played.
Is there any option in asterisk to increase the volume of the IVR files or
On Nov 19, 2007 10:13 AM, Atis Lezdins wrote:
Tilghman Lesher wrote:
On Monday 19 November 2007 02:45:17 Abdul wrote:
Hi all,
I was playing with asterisk .gsm sound file to work for callback.
But the quality is very poor and sound is very low so we cannot clearly
hear what is
On Nov 19, 2007 9:59 PM, Bill Binko wrote:
Hello, everyone
I'm relatively new to Asterisk (and VOIP in general), but I have a
project that it will really help with. So, I setup a test system on an
ancient 400MHz P3 we had lying around. It worked great. I had a test
dialplan working,
http://lists.digium.com/mailman/listinfo/
[...]
If you are having trouble using the lists, please contact
[EMAIL PROTECTED]
--
On Nov 18, 2007 10:43 PM, Philip Prindeville wrote:
Yeah, I posted several hours ago and I haven't seen mine either.
-Philip
On Nov 18, 2007 11:13 PM, Baji Panchumarti wrote:
http://lists.digium.com/mailman/listinfo/
If you are having trouble using the lists, please contact
[EMAIL PROTECTED]
Clicking on the link for asterisk-users takes you to :
http://lists.digium.com
http://www.voip-info.org/wiki/view/Asterisk+sound+files+international
http://voicevector.com/
http://www.voip-info.org/wiki/view/Asterisk+sound+files
I recall running into really great sounding French recordings
somewhere, hope you find the info and sounds in the above.
-baji.
--
On
On Nov 17, 2007 Brian J. Murrell wrote:
[...] which means having to buy something like what
I have and spoofing a little to make it do what I want.
just out of curiosity are you dialing out of * on an analog
line or are you terminating thru a sip provider, or
some other way ?
--
On Nov 17, 2007 8:30 PM, Brian J. Murrell wrote:
Both for now. I am in transition from analog to a VSP (via IAX in
fact).
I know where you are going with this, yes, it seems my VSP will
take a 1NXXNXX even for numbers which are local, and
we are in 10 digit dialling land here now,
You came thru this time.
On Nov 17, 2007 11:06 PM, Jesse Molina [EMAIL PROTECTED] wrote:
I'm trying this again because the last attempt didn't go through (thus
more or less proving one of the below to be true.)
Jesse Molina wrote:
Test123
My messages to this mailing list are
I know this is a lame suggestion, but worth a shot if uninterrupted
participation on this list is important to you, get a gmail address.
gmail removes source IP info, effectively making their server
address the src IP of the msg. Since @gmail is the most popular
domain here, any problem
On Nov 15, 2007 11:01 AM, Steve Totaro wrote:
[...]
I may be wrong, wouldn't be the first time but I think you
need to buy ABE to use Dialogic boards. If that is not
correct, someone please correct me.
thanks steve, I didn't know that they were supported.
I looked in the
On Nov 15, 2007 12:32 PM, Philipp Kempgen wrote:
Last message received at
2007-11-14 18:02:04 GMT
not the case boss, there has been steady traffic for the
past 6 hours, I estimate around 20 msgs, mostly replies.
check your incoming server.
--
I have no idea if this would work :
exten = _0111NXXNXX,1,Set(x=${EXTEN:4})
exten = _0111NXXNXX,n,Goto(${x},1)
exten = _0111NXXNXX,n,NoOp( Sorry it didn't work ! )
exten = _0111NXXNXX,n,Hangup()
;
exten = _NXXNXX,1,NoOp( OMG, It worked ! )
exten = _NXXNXX,n,NoOp( continue
On Nov 14, 2007 12:52 AM, Erik Anderson wrote:
On Nov 13, 2007 11:44 PM, Mohammad Shokuie wrote:
HI Erik,
thanks for your post, Actually im sending new posts not
replying but if you see them correct, how come its wrongly
viewed for me. Are you using a speciall software to view
On Nov 14, 2007 6:31 PM, joakimsen wrote:
You need some experiance with the ANSI C programming language.
Once you have acquired that the rest is pretty straightforward.
http://www.amazon.com/C-Programming-Language-2nd-Ed/dp/0131103709/
--
___
Gary wrote:
I used ChanSpy( ) recorded some test conversations.
It has .raw extension.
What kind of audio file is this? How can I play it?
I don't know, but you can import a raw audio file into
audacity making different parameter selections,
eg. sampling rate ( 8khz ) and format ( ulaw,
On Nov 13, 2007 3:13 PM, Zaheer K. Master wrote:
OK Thanks!
If I'm building a new Asterisk system from scratch, is
there any downside to using CentOS 4 instead of 5?
some of the packages in 4 (4.5) are pretty old, eg. sox
that is bundled with 4.5 does not recognize several
common audio
On Nov 13, 2007 4:30 PM, Kyle Sexton wrote:
[...]
The nice thing about CentOS (as opposed to Redhat proper) is
that they provide the CentOS Plus repository, so installing
PHP5/MySQL would be something like:
# yum --enablerepo=centosplus install php php-mysql
Amen ! I have used
On Nov 13, 2007 Jonn R Taylor wrote:
[...]
The other thing with CentOS vs Debian is that CentOS packages
do not change every month or so. Debain seems to a little more
on the bleeding edge of this, which is not the best thing for a
production system. It is totally person preference
On Nov 14, 2007 12:21 AM, Mohammad Shokuie wrote:
Hi all,
Anyone knows what is wrong with this mailing list its a while all
my new posts appear as a reply (branch) for others post, is
there any hints i could prevent this issue??
Regards.
not this time, came thru fine.
On Nov 14, 2007 12:52 AM, Eric ManxPower Wieling wrote:
[...]
Generally people that experience this problem either have
overly aggressive spam filters or they are sending from an
address different from the one the subscribed from.
he has a hotmail address, my money is on their bulk mail
On Nov 11, 2007 12:40 AM, Vincent wrote:
Thanks, but it won't do, as I need to get the exact filename so
I can send an e-mail pointing to the file later in the script :-/
you can generate your own name using a combo of
STRFTIME() CALLERID() CDR() ( and RAND() if you like )
On Nov 11, 2007 1:07 PM, Steve Totaro wrote:
[...]
Your company (Xorcom) is a direct competitor of Sangoma,
is that correct? I rarely answer questions or give competitors
ideas that may come back and hurt my business.
with all due respect, you are being presumptive projecting
your
(clarification)
On Nov 11, 2007 1:07 PM, Steve Totaro wrote:
[...]
Your company (Xorcom) is a direct competitor of Sangoma,
is that correct? I rarely answer questions or give competitors
ideas that may come back and hurt my business.
with all due respect, you are being presumptive
On Nov 10, 2007 4:12 AM, Mark Quitoriano wrote:
Hi im using centos 5 what is the prerequisite to be installed
before compilling asterisk 1.4?
Thanks!
centos 5 (as well as 4.5) have a lot of extras geared towards
the enterprise, and that is fine if your server is running * and
a bunch of
On Nov 10, 2007 7:34 PM, Vincent wrote:
[...]
exten = _[1-4],n,TrySystem(mv /tmp/msg%d.wav /var/www/asterisk/)
[...]
TrySystem is passing the cmd to (bash) shell, just give it a file match
skeleton as long as you don't have other msgNNN.wav files that
shouldn't be moved.
so
exten =
could it be Return() is missing the parenthesis.
On Nov 9, 2007 8:07 AM, Vincent wrote:
Hello
Instead of using PrivacyManager, I'd rather use my own
dialplan to prompt the user for a ten-digit number if they called
while blocking CID.
This code does prompt the user, but
1)
On Nov 9, 2007 2:53 PM, bilal ghayyad wrote:
[...]
From the other side, I think that baji is talking about
something else than the IP Trunk, he is talking
about outbound [...]
correct, I was responding to Gabriel's post on being
registered w/ SIP provider accepting inbound, but
having
On Nov 7, 2007 9:16 AM, Luis Antonio Prata Barbosa wrote:
Hi,
About APs I think DD-WRT firmware is a very good option, and it
could be used in some versions of WRT-54G...
That's all.
as far as QoS for * goes, Tomato offers a very easy to use tool
that works really well, I am able
On Nov 7, 2007 11:03 AM, Lutgring, Sam wrote:
Thanks for the suggestion. Everything was there except for the context
in the Pickup()cmd and that did not fix it.
Watching the cli in debug you can see it dial the **212 and fall
straight through the first step exten =
On Nov 7, 2007 7:55 PM, Eric ManxPower Wieling wrote:
Baji Panchumarti wrote:
what happens if you replace the pattern matching expr with _.XXX
Not what you expect, that is for sure! The . pattern MUST be the LAST
character in the pattern. Once Asterisk sees a . in a pattern it
stops
after a copious loss of follicles :-), I finally got outbound working.
Basically the channel statement in the call file needs to have the
number to be called. For eg., in test.call format the statement
as follows :
Channel: SIP/3012345678@your-sip-provider
And there is no need for a
http://nooss.org/wiki/Installing_Asterisk_on_Debian_From_Source
Thank you everyone for all your help.
--
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On 10/31/07, Douglas Garstang wrote:
I guess... it shouldn't be too hard to find the time out value
in the source and change it
I couldn't find any timeout related parameter in
app_addon_sql_mysql.c
You may find a default value in one of the header files.
I am wondering if
On 10/31/07, Doug Lytle wrote:
Excellent!
Thank you both!
Doug
don't forget that line of code will disappear the next time
you upgrade your * addons, unless the change makes
it into the official code base.
--
___
--Bandwidth and
On 10/31/07, randulo wrote:
May be of interest to you:
http://www.blogtv.com/Shows/96/YeTrZe3vb2Vpos=ancr
not long enough !
Hot tip : you can skip the first 6 minutes
thnx !
--
___
--Bandwidth and Colocation Provided by
Jason,
I think there is a bit of terminology confusion here,
you can easily convert from format to another using
sox, so if your * server is going to record and email
you a voicemail file, it can surely sox the file to whatever
format the iphone needs it in and then send the email.
It
On 10/22/07, Il Neofita wrote:
There is a way to force the order of the codecs in the sip.conf
since the allow seams to let know only the accepted codec.
I don't know about sip specifically, but from what I recall
reading the .conf files use disallow=all and then add
codecs one by
With all due respect, please try not to make up spellings based
on pronunciation. There is no taff task it is tough task.
If someone is going to take the time to answer a question, the
least we can do is clearly communicate the question.
Spellcheck is readily available where needed.
My
On 10/21/07, Doug Lytle wrote:
Hey everybody,
I've been using mysql databases more and more. I've run across a couple
of instances where I've either made a mistake on the ip address of the
mysql database or for whatever reason, mysql wasn't running. In those
instances, I've noted that
On 10/19/07, Tzafrir Cohen wrote:
On Thu, Oct 18, 2007 at 11:24:24PM -0400, Baji Panchumarti wrote:
I hope you have better success than I did, my problem was
not so much with asterisk in particular but 64-bit in general.
Examples of problems using CentOS 4.5 on x86_64
- many
I hope you have better success than I did, my problem was
not so much with asterisk in particular but 64-bit in general.
Examples of problems using CentOS 4.5 on x86_64
- many problems loading php5 mysql from package
repositories.
- a few asterisk functions don't work, eg STRFTIME()
Zaheer,
this post did show up on the 11th, I am guessing few
people have attempted this, hence no feedback.
-baji.
--
On 10/11/07, Zaheer Master wrote:
Hi All,
I have done some research on Asterisk and I would like to try it in my
office. Here's what I'm looking at for my system:
On 10/15/07, Mojo wrote:
Alan Lord wrote:
Can I do this?
I have a x100p card on my PSTN line and I have an incoming context for
these calls which uses the s extension. I'm wanting to set up a simple
IVR and would like to be able to test the dialplan as I go. But having
to dial-in
On 10/13/07, Turbo Fredriksson [EMAIL PROTECTED] wrote:
Setting debug shows that it skipps the 's' parts...
Why?
because you don't seem to have
exten = s,1,something.
--
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
On 10/12/07, D4rk F1ber wrote:
So I have my asterisk box up and working internally at home and all is
good so far. The next thing I wanted to do was make and recieve calls
to regular land lines now.
I don't have a POTS line and was looking for probably a SIP trunk. [...]
There are
On 10/12/07, Mojo wrote:
Mojo wrote:
Steve Edwards wrote:
You can use IAX or SIP with either provider.
Just realized though, they claim there are audio problems
when using their servers via the iax protocol. They
recommend you use SIP to connect to them.
Should have
On 10/11/07, Dominic Son wrote:
Hi. I'd like for my sound files to be exposed through http.
You know, the ones located in var/lib/asterisk/sounds.
This is probably an apache thing i'd have to configure or is
accessible through some asterisk http routing?
1. how one would configure this?
On 10/9/07, Brian West wrote:
[...] All I did was click edit in frontpage and alert them
of anonymous publishing priv. were on their servers
and they called the FBI [...]
I believe you.
The astonishing security holes that were engineered
by MS so their web
On 10/9/07, Gregory Machin wrote:
Hi
if i want to use sip client to connect to my asterisk pbx do i need to
run a sip server ?
If so can you point me in the direction of a good howto for asterisk and sip
...
install any sip client on your workstation computer
and point it to your
On 10/9/07, Brian West wrote:
http://www.imagestream.com/PCI_921-CDS.html
[...]
off-topic :
I saw Imagestream at the Ohio Linuxfest a weekend ago.
Also picked up a few literature bags by Digium :-)
--
___
--Bandwidth and Colocation
On 10/9/07, Matt wrote:
http://www.usdoj.gov/criminal/cybercrime/WestPlea.htm
Hey,
I am not sure what your point is, are you trying to shame
West on this list with your post ?
He is a contributor to the asterisk movement, which is the
purpose of these lists.
This was uncalled for.
On 10/8/07, Tilghman Lesher wrote:
On Sunday 07 October 2007 20:56, Baji Panchumarti wrote:
could this be the reason for my problem ?
( I am using a 64 bit AMD processor )
2007-09-12 20:12 + [r82285] Tilghman Lesher [EMAIL PROTECTED]
* main/stdtime/private.h, main
On 10/8/07, Tilghman Lesher wrote:
No such board was ever announced. There were rumors of such
a board, but nothing ever got past rumors.
I wasn't into * back then, but apparently they did, for
clarification (I know this not proof, but it is unlikely that
all these people imagined such
.
--
On 10/7/07, Baji Panchumarti [EMAIL PROTECTED] wrote:
hello,
running asterisk 1.4.11 on CentOS 4.5
I am getting no response on function STRPTIME() the system just hangs,
STRFTIME() is working fine as seen below. Same thing happens whether
I called in from a softphone or via
hello,
running asterisk 1.4.11 on CentOS 4.5
I am getting no response on function STRPTIME() the system just hangs,
STRFTIME() is working fine as seen below. Same thing happens whether
I called in from a softphone or via teliax.
While executing the following code :
;
exten =
could this be the reason for my problem ?
( I am using a 64 bit AMD processor )
2007-09-12 20:12 + [r82285] Tilghman Lesher [EMAIL PROTECTED]
* main/stdtime/private.h, main/stdtime/tzfile.h,
include/asterisk/localtime.h, main/stdtime/localtime.c: Working
on issue #10531
On 10/7/07, Justin Case wrote:
Hi List,
I am trying to learn SIP in its entirety. I have so far found:
http://www.amazon.com/SIP-Demystified-Gonzalo-Camarillo/dp/0071373403
Anyone know of any other books that are worth reading ?
http://www.geocities.com/intro_to_multimedia/books.html
:
Telling someone to read the RFC bah.. might as well give them a
blanket and pillow because they will fall asleep. chan_sip is just
ugly in every way.
/b
On Oct 7, 2007, at 9:26 PM, Baji Panchumarti wrote:
http://www.faqs.org/rfcs/rfc3261.html
as well as the source in asterisk
On 8/5/07, Michael Joyner wrote:
ᎣᏏᏲ,
I am looking for VOIP (SIP/IAX) providers that support sending
me RDNIS info on forwarded calls. Are there any providers out
there that support this?
I have a hunch that les.net may offer this in their service
On 8/4/07, Matt wrote:
This may sound stupid.. so bear with me for a moment.
Assuming the only access I have to a machine is through asterisk -rx
can I use the ! command?
asterisk -rx help
includes the ! command, but I can't seem to get it to work ie:
asterisk -rx ! ls
Any help?
On 8/3/07, bilal ghayyad wrote:
Hi List;
What is the difference between WaitExten function and
TIMEOUT (response)? As I see that both are used to
determine the allowed time to enter the digits, any
one can advise?
core show function TIMEOUT
for different timeout parameters, I haven't
On 8/2/07, John Meksavan wrote:
Asterisk Users,
I recently ran into some problems with the quality of service with Teliax.
This occurred on August 1, 2007 with a dropped outbound call, audio
quality isse on the callee side- not hearing me well on callee side, and
sending DTMF tones
On 8/1/07, hugolivude wrote:
Hi,
I'm installing * 1.4.9 and a couple questions have come up:
1) I read here (
http://www.voip-info.org/wiki/view/Asterisk+installation+for+CentOS+4.x)
that if you are using E1 cards you need to install LIBPRI. I'm not using
any cards on this system, so
On 7/30/07, Jared Smith wrote:
Just for your information, IAX traffic is UDP, not TCP. I just thought
I'd bring that up so that someone didn't mistakenly open up their
firewall for TCP traffic instead of UDP traffic and wonder why IAX
traffic wasn't making it through.
Amen !
I had
On 7/27/07, Jared Smith wrote:
Yes, the second edition of the book will be out very soon now. I'm glad
to hear you enjoyed the book. Hopefully you'll like the second edition
even better. :-)
I am sure I will, I am also glad to see that there is someone like O'Reilly
out there
On 7/27/07, Jared Smith wrote:
[...]
As a second suggestion, I'll put in a shameless plug for the many
different Asterisk books on the market. (Yes, I admit it, I helped
author one of them.) Many of them have reference sections that are
indexed and list the dialplan applications
On 7/27/07, Matt [EMAIL PROTECTED] wrote:
Can someone comfirm my logic here?
If I want a phone to use G729 I can set it to use G729... do I
also need to set it in Asterisk? I'm thinking no... as long as
asterisk WILL do G729... if that's all the device accepts it should go
to that
On 7/27/07, Andrea Spadaccini wrote:
Ciao Baji,
[...]
As a second suggestion, I'll put in a shameless plug for the many
different Asterisk books on the market. (Yes, I admit it, I helped
author one of them.) Many of them have reference sections that are
indexed and list the
On 7/25/07, Jaswinder Singh wrote:
Idefisk/zoiper softphone is for IAX2 and it works fine almost
everytime . However there is more variety in sip softphones .
I think zoiper is much better than other iax2 softphones .
Feature wise you are quite right that Zoiper is pretty neat.
But
On 7/26/07, Matt Hoppes wrote:
I would agree... intended to send that to biz, sorry.
I see that you also sent it to the biz-list.
And if you fail the lie detector test how about agreeing to a
full boycott of your service or at least a M.L.D.P. (mailing
list death penalty :-) ?
--
what if your internet provider is blocking inbound 4569 ?
--
On 7/26/07, Michael Munger wrote:
Dear All:
I have several boxes that up and running just great, then we changed
internet equipment due to a lightning strike, now all my inbound IAX
connections (iax2 show peers) have
Marc,
I like your MediaX Phone ( IAX softphone ), I have been using
IDEFISK (Zoiper), but I found your softphone easier to configure.
It is stable and simpler to use.
Keep up the good work.
-baji.
--
On 7/21/07, Time Bandit wrote:
So I am looking for a softphone thats really
SIP has a lot of issues with NAT, I can only get it to work correctly
on my LAN with a softphone.
IDEFISK, now known as Zoiper, is IAX based and I have tested it
from all kinds of hotel rooms, even the free version supports
6 simultaneous calls :
On 7/21/07, Time Bandit [EMAIL PROTECTED] wrote:
You don't specify if he's on Windows, Linux or OSX. But if he is on
Windows, you can try my softphone :
http://www.marccharbonneau.com/asterisk/mediaxphone.php
There is a version using INI file, so you can put all the settings
then zip it
On 7/21/07, Zeeshan Zakaria wrote:
I want my freedom to setup and configure PBX hardware and software
how i want, not how Digium or anybody else wants, so not interested in
Asterisk Appliances.
So anybody with experience with Supply Logics computers. Or any
other recommendations for
On 7/21/07, Andrew Joakimsen wrote:
Check this out: http://www.kapanga.net/IP/home.cfm
Very easy to create a self-installing pre-configured soft phone.
I don't see IAX listed as one of the features, do you know
if it is supported ?
http://www.kapanga.net/IP/specs.cfm
--
I am able to use sox to convert audio files from ulaw to
wav (MS ADPCM), is there a way, using sox or another
command line tool, to convert them to g726 ?
( g726-32 since it is supported by * )
tia,
-baji.
--
___
--Bandwidth and Colocation
On 7/20/07, Thomas Kenyon wrote:
convert file.g729 file.g726-32 in the asterisk CLI works here.
as does file.g726-16 (but not 24 or 40).
The weird thing is, it doesn't seem to transcode from ulaw/alaw but
works fine from g729/gsm.
thank you ! now I have another command to experiment
On 7/20/07, I wrote:
On 7/20/07, Thomas Kenyon wrote:
convert file.g729 file.g726-32 in the asterisk CLI works here.
as does file.g726-16 (but not 24 or 40).
The weird thing is, it doesn't seem to transcode from ulaw/alaw but
works fine from g729/gsm.
thank you ! now I have
The page has been wiki-fied and looks more usable, thank
Mat Kovach of NOOSS for the suggestion and enhancements.
http://nooss.org/wiki/Installing_Asterisk_From_Source
thnx,
-baji.
--
On 7/7/07, Baji Panchumarti wrote:
Just a quick listing of tested, and updated, steps from my
On 7/8/07, Dovid B wrote:
[...]
I was actually thinking of creating a script that you download and it preps
your system for an asterisk install and it does everything for you. It can
also have an option to run as a cron job and update nightly. The issue is
that you cant just update some
Just a quick listing of tested, and updated, steps from my notes.
Enjoy !
http://asterisk-notes.blogspot.com/2007/07/installing-asterisk-from-source-on.html
-baji.
--
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
.
Thank you again for your comments and feedback.
-baji.
--
On 7/7/07, Tzafrir Cohen wrote:
On Sat, Jul 07, 2007 at 12:02:41PM -0400, Baji Panchumarti wrote:
Just a quick listing of tested, and updated, steps from my notes.
Enjoy !
http://asterisk-notes.blogspot.com/2007/07
On 7/7/07, Alex Roston wrote:
I'm in the US. California, specifically.
I guess the other question is what is your need ?
Inbound (need a DID), outbound only (don't need a DID)
or both in outbound ?
Teliax charges $4.95 / mo / DID and 2cents /min for in out
if you use pay-as-you-go.
On 7/7/07, Tzafrir Cohen wrote:
On Sat, Jul 07, 2007 at 12:02:41PM -0400, Baji Panchumarti wrote:
Just a quick listing of tested, and updated, steps from my notes.
Enjoy !
http://asterisk-notes.blogspot.com/2007/07/installing-asterisk-from-source-on.html
[...]
3. The process
On 4/17/07, Salvatore Giudice wrote:
(sorry about the repost. I accidently had an unrelated
subject in the original)
Can anyone recommend a VoIP provider who supports LNP?
I need to move to a new provider for inbound calling and I
want to keep my current numbers. My current provider is a
On 4/17/07, Carlos Jerónimo wrote:
HI, my sip.conf /codecs
disallow=all
allow=ulaw
allow=alaw
this codcs is correct?
thanks
it can be if you have sound files with one of the following
extensions :
au / alaw / al / pcm / ulaw / ul / mu
if you have .sln or .wav files then you are not
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