But apart from that: have you tried at least building that driver with
1.4.0 ?
Yep. The build process seems to work just fine. The ztcfg and zttool
stuff all acts normal. I copied tor2.c and tor2-hw.h from the custom
1.4.0-beta1 drivers (that work just fine with asterisk 1.4.0-beta1) and
Guys, anybody knows if 1.4 has support for unicall or if/which version of
unicall will compile on it?
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Sounds great. What's the mechanism by which Asterisk servers communicate the
mwi status between them?
-Original Message-
From: Jean-Yves Avenard [mailto:[EMAIL PROTECTED]
Sent: Monday, December 25, 2006 11:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hello,
I am working with a php/agi example now and really don't like the way flight
sounds...I am just using it like below. Is there a better voice app to
use? Also, I am wanting the agi to hit a webservice so it will return an
array, is it possible to have asterisk read the array and allow
I've loaded Asterisk 1.4 with the addons 1.4, libpri 1.4 and Zaptel 1.4
as well. I can place calls but I noticed the MySQL was writing out to
the database. When doing an Asterisk load with asterisk - I saw the
following:
[Dec 26 11:02:08] WARNING[10029]: loader.c:375 load_dynamic_module:
Savoy, Kevin - Williston, ND wrote:
I’ve loaded Asterisk 1.4 with the addons 1.4, libpri 1.4 and Zaptel 1.4
as well. I can place calls but I noticed the MySQL was writing out to
the database. When doing an Asterisk load with asterisk – I saw the
following:
[Dec 26 11:02:08]
blackwater dev wrote:
Hello,
I am working with a php/agi example now and really don't like the way
flight sounds...I am just using it like below. Is there a better voice
app to use? Also, I am wanting the agi to hit a webservice so it will
return an array, is it possible to have asterisk
Well, this is weird.
After receiving a sip subscribe message from peer 2529266, here's what Asterisk
responds with:
-- (14 headers 0 lines)---
Found user '2529266'
Looking for 2943110 in bell_CallStart (domain ua2.ipt.xxx.com)
Dec 26 10:19:34 NOTICE[27345]: pbx.c:1741 pbx_extension_helper:
Well there's ya problem.
If 2943110 doesn't have a match in the dialplan anywhere, Asterisk pukes.
What's up with that? I don't see why that is necessary.
Doug.
-Original Message-
From: Douglas Garstang
Sent: Tuesday, December 26, 2006 10:25 AM
To: Asterisk Users Mailing List -
Is there anything available for contolled playback of mp3 files that
offers the same functionality(rewind,skip,pause) as the controlplayback
command does for gsm and wav?
Thanks,
Mike Clark
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Douglas Garstang wrote:
Well there's ya problem.
If 2943110 doesn't have a match in the dialplan anywhere, Asterisk
pukes. What's up with that? I don't see why that is necessary.
Doug.
I'm slightly confused by what you mean... can you elaborate more?
--
Joshua Colp
Software Developer
How to limit the duration of the MeetMe conversation?
--
thank you
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Is anybody else having problems with Asterisk 1.4.0 with the G.729 codec?
I am running on a Dual PIII-866 and have tried the i386, i586, i686 and
pentium3m (not in that order) versions of the module.
Whenever a call comes in that needs to be transcoded, there is no audio
for a couple of
To put it generically, if user A subscribes to the status of user B, and there
is no dialplan match for user B, then Asterisk will return 404 Not Found to
user A.
-Original Message-
From: Joshua Colp [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 26, 2006 10:56 AM
To: Asterisk
Today I installed Asterisk 1.4 in my office and I noticed that it is
still missing the vm-youhaveno sound for voicemail. Without this sound
voicemail will not work! This was reported since the first beta and is
still not included. Be careful if you are upgrading.
--
Telecomunicaciones
Carlos Chavez wrote:
Today I installed Asterisk 1.4 in my office and I noticed that it is
still missing the vm-youhaveno sound for voicemail. Without this sound
voicemail will not work! This was reported since the first beta and is
still not included. Be careful if you are upgrading.
On 26/12/06, Douglas Garstang [EMAIL PROTECTED] wrote:
To put it generically, if user A subscribes to the status of user B, and there
is no dialplan match for user B, then Asterisk will return 404 Not Found to
user A.
Yes, because the subscribe is against an extension, which is
translated to
On Tue, 2006-12-26 at 14:46 -0400, Joshua Colp wrote:
Carlos Chavez wrote:
Today I installed Asterisk 1.4 in my office and I noticed that it is
still missing the vm-youhaveno sound for voicemail. Without this sound
voicemail will not work! This was reported since the first beta and is
I just got cepstal working fine in the dial plan using code like:
exten = 511,5,AGI(cepstral.pl|Welcome to my house finder. At the beep
enter your zip code.)
The php script it calls is based on the nerdvittles weather one so it calls
a webpage which prints to the screen, the nerdvittles code
Even when I move my license to the new install, I got no G729 license
available on the new system.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas Kenyon
Sent: Tuesday, December 26, 2006 1:32 PM
To: asterisk-users@lists.digium.com
Subject:
Asterisk, imho, should still accept the subscription request from user A.
-Original Message-
From: Peter Bowyer [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 26, 2006 11:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP
Why? You're saying 'please update me on the status of extension
'1234'' when there's no such extension. Where's it going to get the
data from?
Better to get a 404, know something's wrong and correct a typo than
let it succeed and just not work.
Peter
On 26/12/06, Douglas Garstang [EMAIL
All:
I am looking to move cell phone providers. I have acquired the new cell
phone and LOVE my new number but want to keep the old number as well. The
new provider only will allow me to use one number or the other. They will
port the old number if I want, but will keep my new number if I ask
All:
(This time with a subject line!)
I am looking to move cell phone providers. I have acquired the new cell
phone and LOVE my new number but want to keep the old number as well. The
new provider only will allow me to use one number or the other. They will
port the old number if I want,
This is obvious (However not an Asterisk question):
You need to ask your old cell provider to port the number to the new one.
Also, in the meantime you need to ask to forward the number to your new
number.
However, since there is nothing forced to him to do that, this is going to
happen if
Has anyone tried to get 1.4 running with a call server 1000 and SIP?
I had 1.0.X running with a call server 1000 and had to tweek the code
due to multipart SDP headers.
Has multipart SDP headers been enhanced in 1.4.
THanks,
Jerry
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At 09:37 AM 12/25/2006, you wrote:
The Asterisk Development Team is pleased to announce the first
release in the Asterisk 1.4 series, Asterisk 1.4.0!
Another thing I've noticed is that twice today while sitting at the
CLI prompt I was throw to the command line because Asterisk had
exited.
Carlos Alperin wrote:
Even when I move my license to the new install, I got no G729 license
available on the new system.
When you say new system, are you referring to a different computer?
Your license is bound to the machine (well the network cards/MAC
addresses) that it is registered to.
No, I didn't said a new machine.
I have a license with 23 active channels for G.729 for testing purposes.
I only erase the old configuration before recompile it, since I was doing
tests with the beta-2 release where the license was working.
So, the only thing I didn't delete was the /licenses
Hi, All
How do I install Zaptel drivers on a system running Suse?
Make results:
grep: /lib/modules/2.6.16.13-4-smp/build/include/linux/autoconf.h: No existe
el fichero o el directorio
make[1]: Entering directory /usr/src/asterisk/zaptel/zaptel-1.4.0-beta2
make -C
This has been fixed in 1.4.0 - I would strongly recommend using that instead of
the beta.
- Original Message -
From: Marco Torrez [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, December 26, 2006 5:44:41 PM GMT-0600 US/Central
Subject: [asterisk-users] I cant
You need the kernel source installed to compile Zaptel.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marco
Torrez
Sent: Tuesday, December 26, 2006 4:45 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] I cant install zaptel
Why don't you try app_swift?
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Swift
This one even compiles on 1.4, and has buffering, meaning that it
doesn't have to wait for the tts to generate the complete output.
http://www.loopfree.net/app_swift/
exten = s,1,AGI(getinfo.php)
exten
Carlos Alperin wrote:
No, I didn't said a new machine.
Sorry, misunderstanding, when you said new system I read that as new
machine.
I have a license with 23 active channels for G.729 for testing purposes.
I only erase the old configuration before recompile it, since I was doing
tests with
The RealTime command pulls all the entire record from the database and
prepends all the fields with the last argument (here is have DN_) so
when the record is pulled, all the records info is available as a
variable like DN_port and DN_ipaddr.
This is a really cool command. Hope this helps.
Does anyone have personal experience with a 5.8gig wireless phone (system)
that has an MWI that WORKS with asterisk via fxs (in my case spa3k)
generated MWI. I know the spa3k does stuttered dialtone but not sure if it
generates FSK MWI.
Uniden DSS7815 MWI works with SPA3K.
Cameron
Actually, there was recently a bug fixed regarding multipart SDP parsing in
chan_sip. That should have fixed the issue with CS1000s and SIP (among other
things). I haven't actually tried it yet on my CS1000, but it should work.
Regards,
- Brad
From: [EMAIL
Hi
On 12/27/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Sounds great. What's the mechanism by which Asterisk servers communicate the
mwi status between them?
With new IAX commands. The client can ask the server how many messages
are waiting.
I've started to port the modification on 1.4,
Too bad Cepstral hasnt still made a decent Spanish voice, the ones they
have still sound too computer like, not like the English ones they have
which sound great!
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Julian J. M.
|Sent:
Lorell Hathcock wrote:
I am looking to move cell phone providers. I have acquired the new cell
phone and LOVE my new number but want to keep the old number as well.
The new provider only will allow me to use one number or the other.
They will port the old number if I want, but will keep my
Hi guys!
We have a call centre that has been moved across from an old Ericsson
MD110 PABX to an Asterisk server with those in the call centre using
X-Lite as their softphone.
I'm trying to get Agent presence configured so that X-Lite gives the
operators a visual indicator of their status -
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