Hello everyone,
A few days ago on *-dev I proposed the idea of making AstLinux images
on a routine basis as a test platform for Asterisk. The ultimate goal
is to have a web driven interface (accessible to the public) where users
can download the latest and greatest versions of Asterisk
Is it practical to 'assume' that in your case mentioned above that
#1 is not going to occur again (since I assume when you say 'line'
you are referring to an outside pstn line), and, #2 is in a mode
of fine-tuning the training when in fact you'd really like it to
start the
On Sun, 28 Aug 2005, Voicomm User wrote:
Hello Group,
Current Setup:
- Eicon Quad BRI ISDN Card.
- 4 ISDN BRI (Telco: Telstra) Onramp2 services.
- Mode: Point2Point.
- 100 Indial Number ranges. Full National Number (9 digit format): BAAXX
where: B (Area code): 2/3/7/8
A (Normal
i need to know something in the zaptel configuration
as it seems i can configure detecting the busy tone and hangup after number of
busy tone counts, that was great but the problem is sometimes the pstn line has
no dialtone and when i try to make call it continue dialing while not having a
On Sat, August 27, 2005 23:41, Michel Koenen said:
Hi all,
I am struggling with the following and I can't get it work:
In the Netherlands where I live it is possible to use special codes
(aka vertical service codes) to set special 'behaviour' of phonecalls.
So e.g. when I want to dial out
On Sun, August 28, 2005 1:15, Aniket Bhat said:
Folks,
I am a newbie to the VOIP world and have a question (might as well
sound silly to some). I would like to set up a PC-to-Phone call from
my desktop to a regular PSTN number. Does the Asterisk PBX itself act
as a VOIP-PSTN gateway or do I
Hi all
i am developing a client for the asterisk that controls ur phone from an Xp c#
application
what functions in Asterisk that will allow you to put someone on hold but not
park calls and bring them back, without using flash hook cuz it will be a
button in that application
Powered by
hello,
According to docs/README.odbcstorage how can we set :
///
The database name (from /etc/asterisk/res_odbc.conf)
is in the
odbcstorage variable in the general section of
voicemail.conf.
You may modify the voicemessages table name by using
Hi all,
I am from India and has been recently using asterisk for testing and enahncing my telephony knowledge. I am trying to use the originate Command from the Asterisk manager on both SIP and ZAP. The command works successfully but does not return any DIALSTATUS such as BUSY,ANSWER,NOANSWER
Hi all,
I am sending the mail again as there was some mistake in the dial plan in the last mail send:
I am from India and has been recently using asterisk for testing and enahncing my telephony knowledge. I am trying to use the originate Command from the Asterisk manager on both SIP and ZAP.
Hello Asterisk-Users,
I copied the speed-dial set at the wiki to my extensions_custom and
included it, the code is:
; Speed dial application. This will store 99 speed dials in the bins 01 - 99
; The database family is called speed and the varible is called spnum
;Storing 11 digit numbers
Please send me a quote for remote installation of
Asterisk, GUI administration, and billing for calling
card, caller ID based prepaid, and postpaid.
Off list please.
Start your day with Yahoo! - make it your home page
Heya,
I'm trying to get SER up and running as a front-end for a couple of Asterisk
boxes for SIP clients. I'd like clients to register with the SER platform.
However, I'd like clients to authenticate with Asterisk when they try to
make outgoing calls via Asterisk. Otherwise it seems that users
hi all asterisk developers and users,
Please help me to configure Astersik with Cisco AS5800
I would like use asterisk for
PSTN(A)- Cisco AS58000 - ASterisk - Audio application
cioa ciao
Start your day with Yahoo! - make
Hello Sean,
Sunday, August 28, 2005, 11:53:28 AM, you wrote:
Hello Asterisk-Users,
I copied the speed-dial set at the wiki to my extensions_custom and
included it, the code is:
; Speed dial application. This will store 99 speed dials in the bins 01 - 99
; The database family is called
Hello Sean,
Sunday, August 28, 2005, 1:38:42 PM, you wrote:
Hello Sean,
Sunday, August 28, 2005, 11:53:28 AM, you wrote:
Hello Asterisk-Users,
I copied the speed-dial set at the wiki to my extensions_custom and
included it, the code is:
; Speed dial application. This will store 99
On Sat, Aug 27, 2005 at 07:41:55PM -0600, Damon Estep wrote:
Does anyone know if gotoiftime can take any subset of 7 days for the
days of the week or only a contiguous range?
According to voip-info.org it has to be one value, a range or '*'.
It is not possible to use a list of values.
Each of
Hello All,
I was wondering if I could do the following on asterisk...
Get a T1 between 2 locations, and split it into a data channel of like
1024, and use the rest for voice channels.
Has anyone done this and had it working well? Or would I need to get a
csu that allows a split into two
I need the quote please. Would appreciate a off list quote.
thanks
On 8/28/05, Chris Felter [EMAIL PROTECTED] wrote:
Please send me a quote for remote installation of
Asterisk, GUI administration, and billing for calling
card, caller ID based prepaid, and postpaid.
Off list please.
[EMAIL PROTECTED] wrote:
Hello All,
I was wondering if I could do the following on asterisk...
Get a T1 between 2 locations, and split it into a data channel of like
1024, and use the rest for voice channels.
Has anyone done this and had it working well? Or would I need to get a
csu that
Is it practical to 'assume' that in your case mentioned above that
#1 is not going to occur again (since I assume when you say 'line'
you are referring to an outside pstn line), and, #2 is in a mode
of fine-tuning the training when in fact you'd really like it to
start the
If this issue exists doesn't it mean that asterisk is unstable anyway?
On Sat, 2005-08-27 at 16:29 -0400, Marc Olivier Chouinard wrote:
I have repeatedly mention this issues, and I keep getting laugh at from
Mark... So I do not think donation to digium will fix the core problem.
Digium
Based on the fine detail you provided my estimate is somewhere between 1
and 10 thousand US dollars.
Vikas wrote:
I need the quote please. Would appreciate a off list quote.
thanks
On 8/28/05, Chris Felter [EMAIL PROTECTED] wrote:
Please send me a quote for remote installation of
Hi
Although canreinvite option is yes,
the asterix doesn't send reinvite and the media is going through the asterix
instead of between the two sip phones.
Both sip phones (handytone 486) are
configure with canreinvite option yes and use the same codec G.729. And
Dial()
command don't
I
have had no issues where asterisk is affected by a Sangoma card being
down.
I
ran my test server like that for a few weeks doing lots of testing before I
brought it up with a dummy card. Even now, if it's up or down it doesn't matter
to asterisk.
Chad
From: [EMAIL PROTECTED]
On Sunday 28 August 2005 10:21, Rich Adamson wrote:
Might try playing around with the canceler parameters on the fxs channel.
Since the analog fxs phone is always very close physically, maybe play
with the echotraining (echocancel=32, and other echo parameters) to
see what impact those might
i need to know something in the zaptel configuration
as it seems i can configure detecting the busy tone and hangup after number
of busy tone
counts, that was great but the problem is sometimes the pstn line has no
dialtone and when i
try to make call it continue dialing while not having a
Might try playing around with the canceler parameters on the fxs channel.
Since the analog fxs phone is always very close physically, maybe play
with the echotraining (echocancel=32, and other echo parameters) to
see what impact those might have. (In theory, using something like
Do you have any other zaptel hardware in
the machine?
Sangoma did confirm this was an issue that
was corrected in beta13 of the the wanpipe drivers.
Asterisk does require a timing source,
either a zaptel card or ztdummy to function correctly.
From:
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Paul
Sent: Sunday, August 28, 2005 7:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] T1 DSU's/Split for voice
[EMAIL PROTECTED]
On Sun, Aug 28, 2005 at 10:45:18AM -, saket setu wrote:
I am trying to use the originate Command from the Asterisk manager on both
SIP and ZAP.
The command works successfully but does not return any DIALSTATUS...
Response: Success
Message: Originate successfully queued
Indeed this
I have not experienced that problem, but earlier firmware resulted in an
unusable speakerphone.
Check if you have the latest firmware, then ask Sipura support for help.
The one time I E-mailed them they were quite responsive.
the 841 still has a worthless display though, doesn't it?
Lack of
* # are valid in a dialplan
you would start your exten = with the vertical service code *21*
then play prompt, collect digits, play prompt, dial
${exten}$(var_for_collected_digits}
BUT, unless I have missed something, You can just send *21* to the PSTN
and then follow their prompts! As long as
bodra wrote:
i need to know something in the zaptel configuration
as it seems i can configure detecting the busy tone and hangup after number of busy tone
counts, that was great but the problem is sometimes the pstn line has no dialtone and
when i try to make call it continue dialing while
Then, on a commercial turn up (back when I did these, it was Western
Union and/or MCI), the tech at the other end would again dialup the
milliwatt, report the value measured over the loop and the pad(s)
re-adjusted to match the values for the loss in a document provided.
That is
sipsak (www.sipsak.org. ) is an excellent tool for
this.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan
Sent: Friday, August 26, 2005
10:48 AM
To: 'Asterisk Developers Mailing
List'
Cc: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Giorgio Incantalupo wrote:
Hi,
is there anybody who knows what this warning means??
WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type
I would bet that [10] doesn't have a type=
___
--Bandwidth and Colocation sponsored by
On Fri, Aug 26, 2005 at 12:31:29PM -0500, Eric Wieling aka ManxPower wrote:
ignorepat does not work for SIP since the dialtone is coming from the
SIP device, not from Asterisk.
You would need to set the phone up to continue dialtone after dialing 9.
Not all phones support that.
Hm. In
Hi!
We are in the process of an Asterisk call center deployment using IAX2
G711 ulaw softphones. Outbound sound quality is terrible.
Have you tried a different sound card and/or a USB handset (which
includes an external sound card)? And what exactly do you mean with
terrible sound?
For those that are interested in the vt1000 paper I wrote a while back,
I have it now on my webpage, at
http://www.0xdecafbad.com/Unlocking-Motorola-VT1000.html
Some of the information there was posted elsewhere, some wasnt.
basically the unit runs vxworks, and it needs a docsis like server to
Hi,
In my office I%u2019m using mixed architecture of Zap and Sip phones,
everything works fine but I have got some problems with picking up Sip
channels. To be certain I can%u2019t do it at all, after I%u2019m dialing *8
the console says nothing to pick up (despite I configure appropriate
Here is my situation. I have
MeetMe conferences going on between internal SIP lines and Zap
channels. I need to be able to join each conference at the beginning
and end, and easily switch between them on request for monitoring. I
also need the option of joining the conference if needed. I have
Ben Brown wrote:
I suppose if there was just a way to monitor the 24 conferences on request, then
the participation could be accomplished using a regular SIP client.
In CVS HEAD (and soon Asterisk 1.2), app_chanspy will do exactly that.
___
Damon Estep wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Paul
Sent: Sunday, August 28, 2005 7:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] T1 DSU's/Split for voice
[EMAIL
Packet8 got around this in an interesting waycharge clients $1.50
per month for E911 or have the option of saying no.
Lol, how many people do you think took them up on that offer?
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
Matt Fredrickson wrote:
On Fri, Aug 26, 2005 at 02:00:54PM -0600, Rich Adamson wrote:
Relative to the fxotune app, it would appear the app is specific
to the v2.4 kernels (/dev/zap*), which the v2.6 kernels don't use
It should with 2.4 and 2.6. 2.6 kernels with properly configured
In my office I%u2019m using mixed architecture of Zap and Sip phones,
everything works
fine but I have got some problems with picking up Sip channels. To be certain I
can%u2019t do
it at all, after I%u2019m dialing *8 the console says nothing to pick up
(despite I
configure appropriate
On Sun, 28 Aug 2005, Kevin P. Fleming wrote:
Ben Brown wrote:
I suppose if there was just a way to monitor the 24 conferences on request,
then the participation could be accomplished using a regular SIP client.
In CVS HEAD (and soon Asterisk 1.2), app_chanspy will do exactly that.
Chanspy
Remarks inline
Dean Collins wrote:
Packet8 got around this in an interesting waycharge clients $1.50
per month for E911 or have the option of saying no.
Lol, how many people do you think took them up on that offer?
From what I understand, Packet8 had this option for quite some time. I
Its the same syntax for every other config. Just look at every other config
option and replicate.
Odbctable=mytablename
Or
Odbctable = mytablename
-Matthew
From: harry gaillac [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello All,
I was wondering if I could do the following on asterisk...
Get a T1 between 2 locations, and split it into a data channel of
like
1024, and use the rest for voice channels.
Has anyone done this and had it working well? Or would I need to
get
a
csu that
I have no problem joining the
conferences and monitoring. What I need is a nice, simple, preferably
GUI method to switch between multiple active connections. I have a
method I like using a 3 line softphone, which works for 3 conferences,
but I need one "line" for each connection to use my
Hi All
Does anyone know if multiple Digium cards on a single machine will be a
problem.
Machine specs: Dual Zeon 3.0GHz on Intel server board.
Cards: TE411P, TDM400P, TDM400P
I will turn off all unnecessary PCI devices; USB, parallel, serial, etc...
Thanks
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Garth van Sittert
Sent: Sunday, August 28, 2005 11:46 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Multiple PCI cards
Hi All
Does anyone know if multiple
There is a patch to mplayer that allows it to suppress stdout messages
and instead output pcm data to stdout. I managed to get it working with
app_mp3.c and seems like it is working fine. All that was needed was a
change in the execl line and a slight increase in timeout value. I have
only done
I am trying to find a way to allow dialout from voicemail when connected
from an 'internal' extension context, but prevent dialout when connected
from an 'external' extension context.
As far as I can tell the dialout context that can be set in voicemail
has no regard for the context from which
I have 2 TE410P's and a TDM400P in same machine without issues
Bart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Damon Estep
Sent: Sunday, August 28, 2005 10:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
On Sat, Aug 27, 2005 at 12:13:43PM -0600, Damon Estep wrote:
Can anyone shed some light on which of these packages are required and
what component requires them? I am in the habit of putting them on, but
in a few cases am not sure if they are still (or were ever) needed.
qt-devel
huh?
Anyone have any experience running an asterisk box with a single nic
and multiple IP's (aliases)?
Have a six class-c production network that needs to be completely
re-IP'ed and need to run the box with both an old and new IP for a few
days.
___
On Sunday 28 August 2005 11:59, Steve Underwood wrote:
I don't follow why knowing that impedance mismatch is the problem has
stopped you making fxotune fix it. :-\ Where you the one who asked me
how to make fxotune work well on IRC? Someone asked a while ago, and
said they were working on a
Thats works without any problems.
Bob.
Dne neděle 28 srpen 2005 21:46 Rich Adamson napsal(a):
Anyone have any experience running an asterisk box with a single nic
and multiple IP's (aliases)?
Have a six class-c production network that needs to be completely
re-IP'ed and need to run the box
Rich Adamson wrote:
Anyone have any experience running an asterisk box with a single nic
and multiple IP's (aliases)?
Have a six class-c production network that needs to be completely
re-IP'ed and need to run the box with both an old and new IP for a few
days.
I'm doing this with just 2
Message: 11
Date: Sun, 28 Aug 2005 11:46:29 +0800
From: chris [EMAIL PROTECTED]
Subject: [Asterisk-Users] error compiling on solaris 10
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain;
Guys. I was checking the changes for 1.2.0 beta1 and I read this:
* Asterisk Realtime Architecture
* Asterisk Manager Interface
* Asterisk Extension Language
* Dialplan functions
* More powerful dialplan expression parser
* Portability enhancements for FreeBSD, OpenBSD,
On 28 Aug 2005 10:35:34 -, saket setu [EMAIL PROTECTED] wrote:
Hi all,
I am from India and has been recently using asterisk for testing and
enahncing my telephony knowledge. I am trying to use the originate Command
from the Asterisk manager on both SIP and ZAP. The command
Go here http://configure.us.dell.com/dellstore/config.aspx?c=uscs=04kc=6W300l=enoc=pe1850-mins=bsd
You can get dual xeon 3Ghz processors and then one
of three other optional upgrades for free.
1 GB RAM or Free Embedded RAID, or free 73 Gig
10KSCSI second HD (comes with one)
From the Digium
Anybody having issues with ztdummy under the current 2.6 RC7? I get the
following errors when trying to modprobe ztdummy:
Unable to register zaptel rtc driver
Doing a Google on the error shows reference to a message from 2004 that
said you might not have RTC compiled into the kernel.
Doug Lytle wrote:
Anybody having issues with ztdummy under the current 2.6 RC7? I get
the following errors when trying to modprobe ztdummy:
Failed to mention that this was under the current Asterisk 1.2 Beta 1
release
___
--Bandwidth and
How many PCI slots? You have to add a PCI
NIC and use 1 of them !!!
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of asterisk
Sent: Sunday, August 28, 2005 5:52
PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Good
http://www.rentacoder.com/RentACoder/misc/BidRequests/ShowBidRequest.asp?lngBidRequestId=325748
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
___
I am assuming two, couldn't a USB NIC be
used? Obviously not gigabit but can anyone see any problems with that
setup?
- Original Message -
From:
Damon Estep
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Sunday, August 28, 2005 4:32
PM
But how many SIP-Zap channels can you have in use simultaneosly? It it a
case of the usage? Does the fact that the machine can handle these devices
also mean that it can use them at the same time, depending on the CPU and
RAM?
- Original Message -
From: Asterisk [EMAIL PROTECTED]
To:
In Admin/Advanced have you tried the Handset Input Gain: settings?
Rob
On 8/28/05, Juan Jose Comellas [EMAIL PROTECTED] wrote:
I have just bought several Sipura SPA-841 SIP phones, and after some testing I
have found out that the volume received by other parties when calling using
the handset
Hi Damon and others,
Your example is still doing what I tried already, so eventually the
dial command ends like:
Dial(zap/4/*21*)
or
Dial(zap/4/*31*)
I prefer to use Dial(zap/4/*21*thenumber)
or Dial(zap/4/*31*thenumber)
But whatever I try, the error message as in my first post shows up and
Hi all,
I was gradually getting to grips with Asterisk at Home and hadgot things workingwith a voipuser account and three extensions (2 X-lites and a Sipura 2100) then suddenly all the extensions went down. Non will login...!
Have not got a clue why. Any ideas?
Also, as a side issue, I set up
On Sun, 2005-08-28 at 21:05 -0400, asterisk wrote:
I am assuming two, couldn't a USB NIC be used? Obviously not gigabit
but can anyone see any problems with that setup?
USB throughput is less than max bandwidth which is what is advertised.
Add a hub and it gets even worse. There is a
Hi Damon and others,
Your example is still doing what I tried already, so eventually the
dial command ends like:
Dial(zap/4/*21*)
or
Dial(zap/4/*31*)
I prefer to use Dial(zap/4/*21*thenumber)
or Dial(zap/4/*31*thenumber)
But whatever I try, the error message as in my first post shows
Hello,
I try set Ua---SERAsterisk (voicemail/ARA)
|
Ua
ser stable
asterisk cvs head
I read
http://mail.iptel.org/pipermail/serusers/2005-February/015997.html
to forward unavailable or busy sip agents to asterisk
voicemail in failure route.
How may I configure
Once upon a time Friday 26 August 2005 12:30 pm, Brian C. Fertig wrote:
Take it from someone who owns 25 of them. Stay away from FC anything.
Use CentOS 4 its better more stable and has true multi-treading as FC
doesn't thread anything..
What do you mean by FC doesnt thread anything?i
You'll want some rules in your sip.conf to handle the connection from
SER. A
starting point might be:
[ser ip addr:ser port ?= 5060]
type=peer
context=my sip context name
tos=lowdelay; tos delay
allow=ulaw ; dtmfmode=inband only works with
Anton Krall wrote:
Also, maybe it was me but I upgraded on a test server by doing the make and
make install over my cvs old one and when on the CLI I do a show version and
I get this:
CVS what? v1-0? HEAD?
server2*CLI show version
Asterisk built by [EMAIL PROTECTED] on a i686 running
Dialtone detection should be an option in .conf for zap channel, i agree
with that.
Are you trying to play with the case where you have an analog phone
bridged on your fxo line, and detect the lack of dialtone when
someone is using that analog phone?
Belive or not, but at some places on the
cmisip wrote:
I want to be able to send a dtmf key to asterisk and have mplayer
forward or rewind.
pabx*CLI show application ControlPlayback
pabx*CLI
-= Info about application 'ControlPlayback' =-
[Synopsis]
Play a file with fast forward and rewind
[Description]
I'd suggest turning off echotraining on the FXS altogether, and perhaps
even
killing the echocanceller on FXS entirely. (you won't be getting
significant
echo from the FXS, and the FXO should be handling it anyway) --
echocancelwhenbridged might be an interesting thing to play with as well.
Since this is a lively topic, I'll ask here...
How can I measure the interval between the original and the echo?
On Mon, 29 Aug 2005, Soner Tari wrote:
I'd suggest turning off echotraining on the FXS altogether, and perhaps
even
killing the echocanceller on FXS entirely. (you won't be
On Sunday 28 August 2005 19:55, Soner Tari wrote:
Andrew sez:
echocancel=64
echocancelwhenbridged=yes
echotraining=800
channel = 1-3
echocancelwhenbridged=no
channel = 4-7
I am sure you know that in zapata.conf parameter settings are in effect
until specifically overridden later
On Mon, 2005-08-29 at 11:37 +1200, Matt Riddell wrote:
cmisip wrote:
I want to be able to send a dtmf key to asterisk and have mplayer
forward or rewind.
pabx*CLI show application ControlPlayback
mplayer has advantages of more codecs as well, so you arent as limited.
In addition it will
Hi Mathew,
We are interested in doing this too, is it possible you can share the
information with us?
We are looking at using a TNT MAX to terminate 8 E1's from the Telco,
but we need a way of receiving the SS7 signalling and passing it to the
TNT's via IPDC or whatever.
Regards,
Andy
Please forgive me, if I misunderstand the problem completely.
Following instructions in several german blogs, I want to configure
Asterisk with a hfc-pci card, an old NTBA and an ISDN phone
as a SIP device.
It seems that I have to set signalling in zapata.conf to bri_net_ptmp.
When I do this,
We have the ability to do this on a large scale, but want to do it on a
smaller scale for 1 to maybe a maximum of 5 TNT's.
Andrew Thrift wrote:
Hi Mathew,
We are interested in doing this too, is it possible you can share the
information with us?
We are looking at using a TNT MAX to
I wrote a php web page to do just this with astguiclient/VICIDIAL
across several servers at once. If you use or want to install
astguiclient let me know and I'll tell you more about how the code
works.
http://astguiclient.sf.net/
MATT---
On 8/28/05, Ben Brown [EMAIL PROTECTED] wrote:
I have
Hi Kevin,
Just courious how to fix the bug for not a blank space in Asterisk
version number? I think I have seen that in the past but downloading a
new copy from CVS fixes it. No harm but it tickles.
David
Kevin P. Fleming wrote:
Anton Krall wrote:
Also, maybe it was me but I upgraded
I upgraded from cvs head 1.0.x which in my case was cvs head about 2 months
ago.
Do you recommend doing a clean install vs. installing on top?
What new dialplan features should I look for and new apps on 1.2.0 beta1?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL
Due to a packaging error, the tarball released on Friday night did not
have a version number embedded in it, which results in various strange
build errors and other odd behavior.
The tarball on the FTP servers has been updated to correct this
situation. Sorry for the inconvenience :-)
Hey,
does anyone know why i'd be receiving:
Aug 28 19:40:04 DEBUG[1875]: # Testing 66.27.233.241 with 10.0.10.0
Aug 28 19:40:04 DEBUG[1875]: Target address 66.27.233.241 is not local,
substituting externip
I get tons of them, usually when the phone is registering/calling/receiving
Hi All,
I currently run asterisk in our office (in Japan) and use a cisco PRI
gateway for connection to the PSTN. I would like to setup some more systems
for our smaller offices (in Japan) that would use BRI and preferably using a
PCI card in the asterisk box and not a seperate Cisco gateway
Anton Krall wrote:
I upgraded from cvs head 1.0.x which in my case was cvs head about 2 months
ago.
There is no such thing as cvs head 1.0.x. You could mean 'CVS v1-0
(whatever was current in the 1.0.x branch at the time) or 'CVS HEAD'
(the current development branch at the time).
Do you
trixter http://www.0xdecafbad.com wrote:
controlplayback seems to fit if all you want is mp3s however ...
Although it works with all supported formats.
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Cheers,
Matt Riddell
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http://www.sineapps.com/news.php (Daily Asterisk News - html)
Adam Robins wrote:
We are in the process of an Asterisk call center deployment using IAX2
G711 ulaw softphones. Outbound sound quality is terrible.
Check if the network card is in half duplex mode.
--
Cheers,
Matt Riddell
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Paul wrote:
Based on the fine detail you provided my estimate is somewhere between 1
and 10 thousand US dollars.
A max of 10K for any job? /me thinks you're selling yourself short.
Just imagine if you got the job and it required 2000 hours.
At $100 per hour that would be $200,000
Even at
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