Damn, I would have greatly enjoyed this... except it would be at least
8pm by the time I get there
Perhaps in a few weeks we should have a sydney version of what happened
in melbourne last week...
PS, I thought about flying to melbourne for the night, and then I woke
up and realised I still
Don't forget to restart your * bos. A simple reload won't work...
David Masure
-Message d'origine-
De : Marco Ziglioli [mailto:[EMAIL PROTECTED]
Envoyé : jeudi 24 février 2005 18:51
À : Asterisk ml post
Objet : [Asterisk-Users] Asterisk and #
Hi ml,
I have a problem related to call
Hi,
In your agents.conf file you just have to add the following entries :
recordagentcalls=yes
recordformat=gsm (or wav,...)
createlinks=yes
savecallsin=/var/spool/... (the directory you want ot use)
Best regards
David Masure
-Message d'origine-
De : Aram Ter-Martirosyan
But is is the same kernel, I asked for the sources to be installed as
part of the config.not sure why it decides to call the kernel
2.6.8.1-12mdk-i586-up-1GB yet dump the sources in 2.6.8.1-12mdk?
I have looked at the kernel rebuild options and looks scary! Maybe this
is a little too much and
Hello ,
I have a question regarding to PRI card (Wildcard TE110P).We want ot use this card in Hungary .So if we have a PRI line (30 B channel (64Kb) and 2 D channel) is is good (I think)
But what happends then when we have only BRI (2-D channel + 1-D channel for signaling) ?
Does it works this
On Fri, Feb 25, 2005 at 11:31:34AM +0930, Hermann Wecke arranged a set of bits
into the following:
Paul A Brown wrote:
Anyone had a Cisco 7970 working with Asterisk?
As 7970 uses SCCP, you can do it with asterisk. I did it with 7960.
Nope, you can't.
As SCCP is not really a protocol, it's
Hi,
it is the same hardware, but with a firmware by Brian F. G. Bidulock.
It has nothing to do with the libisup project, Steve Underwood wrote
several times within this mailing list and soon will be made public
as SS7 support for asterisk with that Digium card.
Roger.
I want to use asterisk dial to PSTN,but only dial,don't connect.
when you hear ring,you only can hungup,don't connect.
when you connect , asterisk will disconnect .
who can tell me what write extension.conf?
Thanks.
___
Asterisk-Users mailing list
Martijn van Oosterhout schrieb:
...
There was also an SS7 status report[2] last June but it's doesn't seem to
have lead anywhere either. There was post saying an SS7 release was
immenent last September[3], but then silence.
Hi,
yes, in the beginning, when we looked for a SS7 solution
for
Hello Jim,
thx for the answer..
Im happy I found someone that is using flash :)
Am I right, if I transfer a call with flash, the line will be free
afterwards ?
Would you mind to past me how you did the flash part @the extention file ?
Also, If I use flash, do I have to setup anything else or
On Fri, 2005-02-25 at 00:14 -0800, asterisk asterisk wrote:
Hello ,
I have a question regarding to PRI card (Wildcard TE110P).
We want ot use this card in Hungary .
So if we have a PRI line (30 B channel (64Kb) and 2 D channel) is is
good (I think)
But what happends then when we have
Does this mean I have to download and re-compile my asterisk sources
inorder to get that file? And if yes, how do I get the sources with cvs
checkout phphconfig? If no, how is it done?
No, only do the cvs checkout phpconfig, and put the files in the right
directory that's all.
Guido Hecken
Hey..
Your saying I can not use flash with ISDN ? What options to I have to
transfer a call directly ? ( So I have a free line afterwords)
What interface are you using? ZapBRI? if so you might be able to do the
hairpinning if it is supported.
Im not using any interface..
But if you know how
This is a bug in asterisk. Caller's exten is saved nowhere, so park cannot call back when timeouts. What you have to do is copy caller's username, as long as it is its extension, to parkee's callee number.
I gave this from SIP's point of view.
[EMAIL PROTECTED] wrote:
Is it only the ATA that has to be T.38 compatible or does Asterisk
have to work with T.38 also? Does Asterisk support T.38?
Asterisk must have T38 support in order to recognize the signaling.
No it doesn't at this time.
We're working on Fax as well and if I'm not
Hi,
I intend to let several SIP-phones on my asterisk ring cascaded on
incoming calls.
First only phone 1 should ring, after 5 seconds phone 2 should ring in
addition and after additional 5 Seconds phone 3 should also ring.
How can I realize that correctly?
Currently I do use
Hallo,
I need to get from sip invite message the sdp block,precisely I need to
know the ip address and port RTP and the codec about the caller.Is there
anyone who can help me?Thank you!
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi everybody,
which version of ast_data I can use for Asterisk v1.0.5?
Regards
Bastian
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
I install WebVmail today on a Fedora 2 box. I got the cgi script running etc
and I get the login prompt. However, when I enter a mailbox and password,
ie. 201 and 1234, I always get a message saying the login is incorrect. Any
tips out there?
Thanks
Martin
On Fri, Feb 25, 2005 at 09:40:00AM +0100, Roger Schreiter wrote:
When I learned about that other project, Steve Underwood
was talking here, I gave up looking after the asterisk-SS7
project by OpenSS7, and begun supporting that libisup project
for asterisk.
You mentioned my very old status
Are you running apache as root or as the asterisk user? If not, maybe
it's a permissions problem...
Julian J. M.
On Fri, 25 Feb 2005 03:39:30 -0600, Martin Keding
[EMAIL PROTECTED] wrote:
I install WebVmail today on a Fedora 2 box. I got the cgi script running etc
and I get the login prompt.
On Fri, 2005-02-25 at 10:32 +0100, Elmar Haneke wrote:
Hi,
I intend to let several SIP-phones on my asterisk ring cascaded on
incoming calls.
First only phone 1 should ring, after 5 seconds phone 2 should ring in
addition and after additional 5 Seconds phone 3 should also ring.
How
Hello,
I'm doing lot of international calls via Sixtel and VoipJet. And there are some calls
which do not go through - Asterisk immediatelly returns with NOANSWER. And it is not
because the dialed party does not pickup the phone, it is because the call does not go
through the provider.
I've
I have configured asterisk with the AMP php configuration utility. I am
able to make outgoing calls through broadvoice but incoming calls are
sent to BV's Voicemail and never actually enter the IVR. When I show
sip debug info through the asterisk prompt it actually reads the
incoming call
has any one implemented asterisk with 723 and 729
codecs, what is the cheapest way.
is there a limitation in the open 723
implementation ??
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hello,
We would like to :
By any web-user (ms explorer) to be able to call from a
web-page to a certain number/extension connected to one specific asterisk.
Almost as a web-based auto-attendant
functionality.
Hence:
surf to the specific web-site
enter the extension
Duane wrote:
For those interested, I'm giving a talk about VoIP/enum.164/asterisk
tonight in Sydney at the Sydney LUG meeting which is about 7pm in the UTS
build #2, 4th floor, room 10.
Sorry for the late notice, it didn't occur to me that there might be
people on this list interested and able to
Hi,
I have postgresql and * all up and running as the latest cvs-250205,
although something weird.
Every outgoing call regardless of whether or not it is answered or busy
or just rings out in the database the entry has the disposition as
ANSWERED, instead of BUSY or NOT ANSWERED.
As a test I
exten = s,1,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED])
[context]
exten = 1,1,Dial(SIP/1)
exten = 2,1,Wait(5)
exten = 2,2,Dial(SIP/2)
exten = 3,1,Wait(10)
exten = 3,2,Dial(SIP/3)
Basically, use the 'local' channel for your dial, then you can wait a
bit before
You would need to somehow have an external fixed address which would
redirect all of your traffic to the dynamic address, I have found no way to
do this, your best bet is to pony up the extra cash for a fixed address
(usually 3-4x the cost)
I would love to hear if anyone has figured this
Pure guess... in the US, probably treated like rtty?
Interesting thought. There's no 'r' part in the 'tty' part in this case,
though (unless you were transmitting rtty through VoIP).
What I sort of meant by rtty was the use-restrictions (content) placed
on the use of radio tty by the FCC
I want to setup a senario in which the callers
hears to some music file while the phone is ringing and as soon as the line is
answered the music is stopped palying. i.e. instead of the rings the caller
listens to some music.
Is is possible with asterisk?
Kindest
Muhammad Muzzamil Luqman
GMRS, FRS and MURS radios may not be interconnected with the PSTN (47
CFR 95.141). There has been a lot of talk from lobbyists to clarify this
rule, but as it stands you could conceivably connect a *private* network
to GMRS or MURS radios (you can't make any plugins or modifications to
James Bean [EMAIL PROTECTED] wrote:
[...]
Every outgoing call regardless of whether or not it is answered or
busy or just rings out in the database the entry has the disposition
as ANSWERED, instead of BUSY or NOT ANSWERED.
As a test I intentionally rang numbers that would be busy or
I thought Vonage did not allow this?
-Randy
Nitesh Divecha wrote:
Hello Asterisk Users,
After Brain storming for couple of hours, days, and weeks, finally got
Asterisk to work with Vonage for Inbound and Outbound calls.
Requirement: -
1) Vonage Softphone account
2) Asterisk
3) Couple of SIP
You could add
exten = 1,2,Goto(context,2,2)
But I don't know what will happen when, after 5 secs, dial SIP/2 is
executed again...
Julian
On Fri, 25 Feb 2005 12:56:14 +0100, Elmar Haneke [EMAIL PROTECTED] wrote:
exten = s,1,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED]Local/[EMAIL
Dial(SIP/whatever,30,m)
instead of 'r'
http://www.voip-info.org/wiki-Asterisk+cmd+dial
Julian
On Fri, 25 Feb 2005 17:18:59 +0500, Muhammad Muzzamil Luqman
[EMAIL PROTECTED] wrote:
I want to setup a senario in which the callers hears to some music file
while the phone is ringing and as
James Bean [EMAIL PROTECTED] wrote:
[...]
Every outgoing call regardless of whether or not it is answered or
busy or just rings out in the database the entry has the
disposition
as ANSWERED, instead of BUSY or NOT ANSWERED.
As a test I intentionally rang numbers that would be busy
What all the world's FAX problems? Even FAX spam? :-)
If you understand what T.38 is you will understand which problems it
addresses (summary: it is important for solving some problems, but
nothing solves them all). Most people who post about T.38 don't actually
have much of a clue about it.
Is there any reason to avoid * on Fedora Core 3 at this time?
Have most/all of the issues been resolved now?
Rich
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
Andreas Sikkema wrote:
[EMAIL PROTECTED] wrote:
Is it only the ATA that has to be T.38 compatible or does Asterisk
have to work with T.38 also? Does Asterisk support T.38?
Asterisk must have T38 support in order to recognize the signaling.
No it doesn't at this time.
We're working
Rich Adamson wrote:
Is there any reason to avoid * on Fedora Core 3 at this time?
Have most/all of the issues been resolved now?
Rich,
Both my Asterisk servers run FC3. The only issue I ran into was the
change in RPMs for the source. FC doesn't distribute the
kernel-source RPM any more.
On Fri, 25 Feb 2005 11:21:46 +0700, milisku
[EMAIL PROTECTED] wrote:
Hi all Iam from indonesia, we want to develop voip using asterisk but
there is reseller product that support asterisk on indonesia. How i can
get Thanks
JOKO PITOYO
sure, http://www.clarisense.co.id/
You were correct Steven - I was picking up the extensions from an
include after a jump !!
Lesson Learned - thanks everyone.
On Thu, 24 Feb 2005 20:18:22 -0600
Steven Critchfield [EMAIL PROTECTED] wrote:
On Thu, 2005-02-24 at 15:49 -0800, Richard J. Sears wrote:
I have a [start] context that
Hi,
I have been doing various testing with asterisk and its been going great.
However I am a bit feedup of using vi for editing configs, and would rather
do it from any machine on my LAN. I am running debian and * via xorcom rapid
on a test PC at the minute.
Hence phpconfig would be great,
On Wed, 2005-02-23 at 14:22 +0100, Roberto Piola wrote:
We have set up an HP DL380 with 3 4BRI cards, Fedora core 2 (kernel 2.6.10)
and asterisk (bristuff-0.2.0-RC7f with asterisk 1.0.5). 4 ports are
configured in TE mode and connected to the PSTN; the other 8 are in NT mode
and connected to
Hello,
Were planning to use Digium cards for eastern
european r2 signalling.
However, we would like to have a few references on the
possibility to realise the signaling.
Please, can anyone tell me whether they have had any success
in this, and if there are any special hook-ups
C. Tomlinson wrote:
I have been doing various testing with asterisk and its been going great.
However I am a bit feedup of using vi for editing configs, and would rather
do it from any machine on my LAN.
Look at WinSCP:
http://www.winscp.org/
which is a lovely program that initially purports to
Hi Terje,
The only East European country my R2 software currently allows for is
teh Czech Republic, since that is the only place I could find
information for. If you have information about the protocol used in
other countries, support should be easy to add.
Regards,
Steve
Terje Myhre wrote:
Mark,
Did you have to make any changes to use the premicell, or was it as simple
as an outgoing landline call?
I am looking into doing this as you can get deals where calls between chosen
numbers are free :-)
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
I tried this but the problem is that on a blind transfer from an outside call, the caller id comes through as the PSTN Callerid and not the transferring extensions. I want the callerid to stay that way, so I guess I'm out of luck at the moment.
On Fri, 2005-02-25 at 02:41, ST wrote:
Richard,
I have been using WinSCP to transfer files across easily without messing
with FTP accounts. I had not found that feature, many thanks for pointing it
out :-D
I will definitely use this from now on until I find a better solution. Do
you have an easy way to reload asterisk after changing
have a look at Quanta.
It has a FISH protocol.
Basically, open the file as
fish://ip.add.re.ss/path/to/file.conf
Edit the file and save.
This is a very nice editor with highlighting for several
languages.
-Herman
On Fri, 2005-02-25 at 15:44, Richard Folwell wrote:
C. Tomlinson wrote:
Steve Underwood,
Would you mind summarizing where/how T.38 functions, and maybe how it
compares to the analog fax environment for the asterisk-users arhives?
Seems to be some misunderstanding, and a lot of interest in handling
faxes in various forms via asterisk. If some these approaches were
If you are using windows, have a look
at Zend Studio that is used for PHP
but can do wonders for other editing apps as well.
-herman
On Fri, 2005-02-25 at 15:52, C. Tomlinson wrote:
Richard,
I have been using WinSCP to transfer files across easily without messing
with FTP accounts. I had
I use FC3 on all our servers including 3 * servers.
I have absolutely no issues what so ever.
You do NOT need the kernel source RPM (which I don't even think exists
anymore) as they've changed how they set up the kernel RPMs somewhere
after FC1.
The source rpm from FC1 (which is a bit old
Richard Folwell wrote:
Look at WinSCP:
snip
It is (almost) worth installing Windows just to be able to use it. :-)
If anyone knows of anything similar that runs under Linux please
enlighten me!
Have a look at the fish io-slave for KDE. Type fish://[EMAIL PROTECTED] in your
Konqueror URL-bar and
Dear all,
I have installed asterisk 1.0.5 on redhat 9
I have installed also, asterisk-oh323 0.6.5 module (successfully
compiled and installed)
Now When I am trying to get asterisk communicate with a Radius (in my
case: it's the VoiceMaster Radius)
I was able to do the following:
After
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, February 24, 2005 10:12 PM
Subject: [Asterisk-Users] Asterisk With Broadvoice
I have configured asterisk with the AMP php configuration utility. I
am able to make outgoing calls through
On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote:
Did you have to make any changes to use the premicell, or was it as simple
as an outgoing landline call?
I am looking into doing this as you can get deals where calls between chosen
numbers are free :-)
Absolutely no changes at all I
I have been doing various testing with asterisk and its been going great.
However I am a bit feedup of using vi for editing configs, and would rather
do it from any machine on my LAN. I am running debian and * via xorcom rapid
on a test PC at the minute.
I had the same problem. So I did a
James Bean [EMAIL PROTECTED] wrote:
[...]
I am sorry I did not see anything in any of the docs about analogue
lines causing ANSWERED response on all calls. Could you point me in
the right direction to a fix or setup that fixes this situation?
The only real fix is to get some form of digital
On 15:04, Fri 25 Feb 05, Eivind Trondsen wrote:
Richard Folwell wrote:
Look at WinSCP:
snip
It is (almost) worth installing Windows just to be able to use it. :-)
If anyone knows of anything similar that runs under Linux please
enlighten me!
scp
This is installed together with the
Can someone explain what this error is?
-- Got SIP response 500 Server Internal Error - Invalid CSEQ number
back from 209.xxx.xxx.xxx
How do I fix this?
.o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
j wrote:
I use FC3 on all our servers including 3 * servers.
Great news.
The regular kernel rpms now come with all the headers and development
stuff included.
You should be able to install the kernel rpm and compile zaptel right
away.
do an rpm -ql kernel | less to check out the contents.
Is there any reason to avoid * on Fedora Core 3 at this time?
Have most/all of the issues been resolved now?
I don't know about the issues on FC3, but I wouldn't want to use a
testing distro on a production server.
If you are looking for a stable distro that cost nothing, have a look
at CentOS
;
[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0
[admin]
secret = secret
;deny=0.0.0.0/0.0.0.0
;permit=209.16.236.73/255.255.255.0
Do this in manger.conf, where xxx.xxx.xxx.0 represents your network:
[admin]
secret = secret
permit=xxx.xxx.xxx.0/255.255.255.0
read =
I am having trouble using cvs, is it possible to use cvsup or any other
method available and still get to install, configure and use phpconfig? If
so, how do I go about it?
Julius.
Does this mean I have to download and re-compile my asterisk sources
inorder to get that file? And if yes, how
On Fri, 2005-02-25 at 08:11 +, Razza wrote:
But is is the same kernel, I asked for the sources to be installed as
part of the config.not sure why it decides to call the kernel
2.6.8.1-12mdk-i586-up-1GB yet dump the sources in 2.6.8.1-12mdk?
On Behalf Of Adam
Goryachev
Sent: 25
Any chance you can share your presentation slides, or handouts etc.
thanks
-Original Message-
From: David Uzzell [mailto:[EMAIL PROTECTED]
Sent: Friday, February 25, 2005 6:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoIP/Asterisk
Hello Everyone,
Just for the record, the E400P-SS7 Is IDENTICAL to Digium's E400P, and
the T400P-SS7 is IDENTICAL to Digium's T400P (NOT Their new
TE410/TE405P). There is no additional 'software' on the card, since all
firmware is uploaded to the Xilinx during driver modprobe.
An E400P-SS7 can
Yes, but the 7970 relies on
the very latest SCCP protocol version which was introduced with Call Manager
4.0. As far as I know the SCCP module for Asterisk doesnt support the
latest SCCP version so I dont think it will work. The 7960 works with
the older versions of SCCP. I have yet to
Hi everybody:
I configured my Asterisk to register to my VoIP provider, and I can make
outgoing calls, but I can't receive any calls with it.
I used Ethereal to sniff the activity of it, and I found something that
might be causing the problem:
When my provider's gateway does the Request: INVITE
This is probably better suited to a cisco forum, but thought i'd drop it
in here also. Using a 7960 with *, and have a very specific need. If the
end user is currently on a call on the 7960, and a new call comes in, i
need the phone to:
show a visual indicator of the call (pref flash the call
Hello
I have yet to discover a software package that would both register and
have ulaw codec. The SIP communicator (Java) came closest to usable,
but didn't have the ulaw codec working. What do you use for
communications?
--
Konrads Smelkovs
Applied IT sorcery.
Would plugging into the headphone jack with a phone-patch-type device
be considered a modification for radios with vox capability?
ah ah so do ' phone-patch-type device' interface
via the to frs/gmrs 2 way radios via the mic jack ?
can someone that know this stuff point out a few urls of the
I found some errors in phpconfig. Open the file cls_phpconfig.php
In the function OC_readConfFile around line 131
change : $this-_OC_the_file[] = fgetc($file);
to : $this-_OC_the_file[] = fgets($file);
In the function OC_checkAccess around line 438
change : $accessFile[] = fgetc($file);
James Bean [EMAIL PROTECTED] wrote:
[...]
I am sorry I did not see anything in any of the docs about analogue
lines causing ANSWERED response on all calls. Could you point me in
the right direction to a fix or setup that fixes this situation?
The only real fix is to get some form of
On Sat, February 26, 2005 1:36, Ronald Hartmann said:
Any chance you can share your presentation slides, or handouts etc.
Sure, but was only slides, no hand outs...
http://www.asterisk.net.au/voip%20in%203%20beers.pdf
http://www.asterisk.net.au/voip%20in%203%20beers.sxi
I am going to try out all the instructions and document it, and then submit
to the wiki so future installations are easier for all :-)
I will post the draft 1st here.
Thanks for the help, lets hope I get it working.
C
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On 2005.02.25 05:53 Rich Adamson wrote:
Steve Underwood,
Would you mind summarizing where/how T.38 functions, and maybe how it
compares to the analog fax environment for the asterisk-users arhives?
I don't mean to speak for Steve, so I hope that Steve will still reply
if he chooses to, but I like
Julius,
I have just setup and installed phpconfig with the help of others on this
mailing list. I didn't use CVS checkout as I don't have CVS installed.
I am about to document the process for the Wiki which I hope will help :)
C
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Vonage doesn't sell just a softphone account -- or at least they didn't
about six months ago when I was a Vonage customer. But they do allow a
softphone as an add-on to an ATA-based account.
Because the softphone account works with openly available soft clients,
it also works with asterisk.
Some time ago, I had the same probs with phpconfig and had to search and
google quite a long time to get it running. Since our systems are now
running fine with phpconfig, I simply forgot the above fgetc/fgets issue.
Therefore...
A wonderful place for all this would be the wiki ;-)
Better
Hi
On Thu, Feb 24, 2005 at 11:41:41AM +0100, Hecken, Guido wrote:
Secondly, is the statement no.2 a line a need to change in a given file?
You have to change/verify some settings in phpconfig_init.php .
Look for fakeuser=admin.
Set $reset_cmd = ./asterisk.reload;
Be shure, the script has
Hmmm,
Looking directly at the .../channels/chan_sip.c code does not get any
clues.
Switch( resp )
...
...
case 480: /* Temporarily Unavailable */
case 404: /* Not Found */
case 410: /* Gone */
case 400: /* Bad Request */
case 500: /* Server error */
case 503:
I have yet to discover a software package that would both register and
have ulaw codec. The SIP communicator (Java) came closest to usable,
but didn't have the ulaw codec working. What do you use for
communications?
for SIP you can use X-Lite :
[EMAIL PROTECTED] wrote:
For T.38 passthrough between RTP channels it doesn't need to know a
great deal. There are some pitfalls, though, due to dumbness
in the T.38
spec.
Are you actually working on this?
Yes, well, with a lot of other things, so progress is erratic. I've
got to solve
On Feb 25, 2005, at 7:55 AM, Steve Underwood wrote:
If you understand what T.38 is you will understand which problems it
addresses (summary: it is important for solving some problems, but
nothing solves them all). Most people who post about T.38 don't
actually have much of a clue about it.
I
Must have missed a few messages :) Vonage always allowed this on
softphone lines. Those are $10/month with metered usage (100 min
included). They also require a hardline (ATA) as the primary line on
the account. It's a working crutch for those folks who need a DID in a
rate-center only vonage
James Bean [EMAIL PROTECTED] wrote:
[...]
Is the kludge done at the software side when the data is pulled out
for accounting and being under say 45 seconds is a no answer or
busy? Or is there a tweak that can be done at the database itself?
Since you're using PostgreSQL, you can use a trigger
There are GMRS radios that support frequency splits... I dont think FRS
does.
-- Mike
- Original Message -
From: TC [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, February 25, 2005 9:10 AM
Subject: Re:
Hi,
I'm not sure the way to change it, but when I d/l it from
http://asterisk.espia-net.net/horde/chora/cvs.php/phpconfig/cls_phpconfig.ph
p?login=2asterisksess=5c8e63576772790cfc2e1dbce354e04d
I had read about the problem with fget's, but presumed this change was the
correct one. However it
The Cheapest way is to purchase 2
licenses, or in multiples of 2 If you need more, from Digium.
You will be beating a dead horse and a
dead carriage and a dead driver if you try to get around G729 licensing. You only
need a license for each answer and originate session that uses g.729
I am developing voicemail and SIP and RAIDUS code for Asterisk Code on
the Fedora Core 3 and having no problems.
I am running on an Intel Pentium 3, 1.5 GHz, mother board stuck inside
an old E-machine case and it is very happy... (I only wish I could find
a Okidata B4250 printer driver or a PCL-6
On Fri, Feb 25, 2005 at 03:15:34PM +0100, Michiel van Baak wrote:
On 15:04, Fri 25 Feb 05, Eivind Trondsen wrote:
Richard Folwell wrote:
Look at WinSCP:
snip
It is (almost) worth installing Windows just to be able to use it. :-)
If anyone knows of anything similar that runs
On Fri, Feb 25, 2005 at 01:52:21PM -, C. Tomlinson wrote:
Richard,
I have been using WinSCP to transfer files across easily without messing
with FTP accounts. I had not found that feature, many thanks for pointing it
out :-D
I will definitely use this from now on until I find a better
For MySQL and other glorified flat-file databases, you would
need to postprocess the data. You may feel more confident
skipping triggers and doing this anyway.
So by that any calls that go out over the net using IAX to
the telco
are considered digital and will report correctly?
I'm asking because I'm planning to install multiple machines from the
same image and I need to know what file(s) I need to backup/restore to
make sure I don't lose my licences in the process. The only options I
can think of are:
- There's a config file, though I've seen no mention of it
- The
Hi all,
How do I config Asterisk so when the directory cmd is used, the name of
the found entry comes from a pre-record gsm file instead of being spelled
letter by letter?
Regards,
Francois
Random Thought:
---
All of us failed to match our dreams of perfection. So I rate us on
1 - 100 of 184 matches
Mail list logo
