21 apr 2009 kl. 11.46 skrev Khaled W. Chehab:
Dears,
When my GW send a call to asterisk v 1.4.24 ,
Asterisk send Status: 420 bad extension (unsupported)
Why? Any modifications should be done one sip.conf
regards
Your gateway is propably requiring a SIP extension Asterisk does not
On Thu, 23 Apr 2009, Rilawich Ango wrote:
Hi all,
I wonder who has the same voice quality problem as what we have.
Below is our configuration.
Company --- asterisk 1.4.22 (g729) --- CISCO --- T1 --- customer
Sometimes, customers told me that they heard our voice not very clear,
like a
Normally, there are 10 concurrent calls in peak. You are right that
usage g729 is due to bandwidth consideration.
On Thu, Apr 23, 2009 at 2:42 PM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Thu, 23 Apr 2009, Rilawich Ango wrote:
Hi all,
I wonder who has the same voice quality
The CM is sending the BYE messages.
Any ideas?
Christian
--- On Wed, 4/22/09, Martin asteriskl...@callthem.info wrote:
From: Martin asteriskl...@callthem.info
Subject: Re: [asterisk-users] Conference problem
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
Is there any method of getting call park working on different numbers for
different customers on the same asterisk server?
Currently running asterisk 1.4.23.1
Cheers!!
___
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Hi,
someone has installed on an Asterisk box (not Trixbox) with Debian Linux,
the HUDlite Server?
Can someone help me in retrieve and install packages???
Thanks all
Marco
___
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Hi,
I have qualify = no .
if I set sip debugging on I can see it - but this gives many long debug
messages.
Is there a way to see the source ip in the cli as the calls scroll up? I only
see the destination ip in the cli .
Tx Shaun___
-- Bandwidth
Hello
For those SOHO customers (ie. at most, a couple of POTS/ISDN
connections and simultaneous SIP calls) who'd rather not use a big,
noisy PC to run Asterisk, I'd like to offer an alternative that has
the following features:
- not old hardware sold on eBay, ie. it must be up-to-date hardware
Hello
For those SOHO customers (ie. at most, a couple of POTS/ISDN
connections and simultaneous SIP calls) who'd rather not use a big,
noisy PC to run Asterisk, I'd like to offer an alternative that has
the following features:
- not old hardware sold on eBay, ie. it must be up-to-date
Hello,
On Thu, 23 Apr 2009 05:18:27 -0500 (CDT), Joe Greco
jgr...@ns.sol.net wrote:
Can you give us some clues as to why you have disqualified the fanless
and/or embedded devices that are normally recommended on the list
(Soekris, etc)?
I haven't: I'd like to know what the options are. I'm
Hi Marco,
Try this: http://yum.trixbox.org/centos/4/RPMS/hudlite-server-1.4.32-1.i386.rpm
Regards
David.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Sambo
Sent: Thursday, 23 April 2009 7:29 PM
To:
On 23 Apr 2009, at 11:34, Vincent wrote:
Hello,
On Thu, 23 Apr 2009 05:18:27 -0500 (CDT), Joe Greco
jgr...@ns.sol.net wrote:
Can you give us some clues as to why you have disqualified the
fanless
and/or embedded devices that are normally recommended on the list
(Soekris, etc)?
I
Hi Guys,
I have a strong feeling the loads on my servers will be shooting up soon...
anyone got any idea how many calls i can expect to put through a
DL360:
Dual Quad Core 2.33ghz
4gb RAM with 1gb allocated for a ramdisk (call recordings)
This server is recording calls (mixmonitor), codec is gsm
On 23 Apr 2009, at 11:34, Vincent wrote:
Hello,
On Thu, 23 Apr 2009 05:18:27 -0500 (CDT), Joe Greco
jgr...@ns.sol.net wrote:
Can you give us some clues as to why you have disqualified the
fanless
and/or embedded devices that are normally recommended on the list
(Soekris, etc)?
On Thu, 23 Apr 2009 11:51:02 +0100, Steve Howes st...@geekinter.net
wrote:
http://tinyurl.com/df8qfm
www.voip-info.org/wiki/view/Asterisk+embedded+systems
Thanks Steve. I knew about this list, but I wanted to make sure there
weren't other, more complete sources about the subject.
So at this
My comment, (forwarded from Bristuff list) - A few people are seeing a
Cause 34 (congestion) from ISDN installs, where there clearly is an
available channel. This was originally related to Bristuff as it
happens to ISDN2 users, but there is at least one report of an
unpatched 1.6.x user seeing the
On 23/04/2009 11:12 p.m., Geraint Lee wrote:
Hi Guys,
I have a strong feeling the loads on my servers will be shooting up
soon... anyone got any idea how many calls i can expect to put through a
DL360:
Dual Quad Core 2.33ghz
4gb RAM with 1gb allocated for a ramdisk (call recordings)
This
So at this point, it seems like it boils down to this:
Soekris
PCEngines
Atcom (IP01: £160+VAT)
Herologic (HL-463: $259)
uCpbx (235.00 EUR)
AstBoxes (168.00 EUR)
Gumstix
HP Thinclient t5720
Probably worth adding the Asus eeeBox to the list. It doesn't have space for a
PCI card and isn't
Well, I see the rpm but my Asterisk box has Debian Linux, and I'm a little
afraid to use alien package to transform rpm to deb. Has HUDlite Server
source?? Like in tar.gz??
2009/4/23 David Klaverstyn d...@klaverstyn.com.au
Hi Marco,
Try this:
You don't say how many SIP registrations you are doing, but I have several
servers with betwen 1000 and 1200 simultaneous registered users 24/7. When we
had the registrations in realtime (cached) with the mysql connector, everything
started failing around 600 users. With the ODBC connector
Hello,
I'm using 1.6.1-rc4.
When sending a userevent, (with UserEvent(MyEvent); in extensions.ael),
I've got :
Event: UserEvent
Privilege: user,all
UserEvent: MyEvent
I can't see any Uniqueid field as mentioned
http://www.voip-info.org/wiki/view/Asterisk+cmd+UserEvent or
No, but as I understand it 1.6 would have that possibility.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of carl Lougher
Sent: Thursday, April 23, 2009 4:54
To: asterisk-users@lists.digium.com
Dears
My scenario is incoming call to asterisk which asterisk in its term will
dial it through its trunk .
I recognized that Asterisk is sending two invites to My Trunk GW IP as you
can see in the debugging below
The first is the default and the second when asterisk receives a 200 OK
Why
So at this point, it seems like it boils down to this:
Soekris
PCEngines
Atcom (IP01: £160+VAT)
Herologic (HL-463: $259)
uCpbx (235.00 EUR)
AstBoxes (168.00 EUR)
Gumstix
HP Thinclient t5720
I recently custom built an Intel Atom 330 Mini-ITX based system for a client
who then took it by
Thanks for that, it's pretty much confirming what i first anticipated... my
intentions are as follows:
agents register with opensips, opensips clusters a set of call recording
servers which then connect to our border servers which will save cdr and
choose the sip/iax provider to send the call to.
I have an issue with one of my installations running Asterisk 1.4.20 that I
need some help with.
Not sure if this is new or not but Zaptel or Libpri is not releasing
channels properly. I have had issues with calls that stay up for eight
hours, long distance on the telco side, so it is more than
I had this problem with a box that I was using Festival tts on.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Thursday, April 23, 2009 9:18 AM
To: Asterisk Users Mailing List -
2009/4/23 Steve Davies davies...@gmail.com:
My comment, (forwarded from Bristuff list) - A few people are seeing a
Cause 34 (congestion) from ISDN installs, where there clearly is an
available channel. This was originally related to Bristuff as it
happens to ISDN2 users, but there is at least
Hi all,
I am new to Asterisk, so I have few questions. I intent to run it wholesale
traffic termination. Apart Asterisk, I want to use Asterist2Billing for
billing. Here are my questions:
-Since my main area of work will be wholesale, is Asterisk2Billing good
solution or you have better
Hi,
i'm currently using Originate command on AMI, i can call a certain
channel like a SIP user SIP/1000 then once 1000 is answered it dials out
to amobile or landline.
Would just like to know if i can use AMI to dialout to a mobile or
landline first (instead of SIP user) and once answered,
Dear Sirs,
I've a Fritz USB 2.1 card like this:
Bus 002 Device 003: ID 057c:1900 AVM GmbH ISDN-Controller FRITZ!Card v2.1
I'd like to use it with trixbox/asterisk.
I've already some experience with other cards and mISDN, but I can't make
this card work with it.
I've downloaded the driver from
Hi All,
I am trying to use the AMD (Answering Machine Detect).
But it is not sending the AMD_Status as either
the Human or Machine, it hangs up in middle.
can any one suggest us, what might be the problem
and possible solution to it.
below is the log
-- Executing AMD(SIP/sip-ffe0, ) in new
On 4/23/09, Sam Hawkin gvrt...@gmail.com wrote:
Hi All,
I am trying to use the AMD (Answering Machine Detect).
But it is not sending the AMD_Status as either
the Human or Machine, it hangs up in middle.
can any one suggest us, what might be the problem
and possible solution to it.
On Thu, 2009-04-23 at 09:18 -0400, Steve Totaro wrote:
Not sure if this is new or not but Zaptel or Libpri is not releasing
channels properly. I have had issues with calls that stay up for
eight hours, long distance on the telco side, so it is more than just
a nuisance.
Have you examined the
Maybe the customer hangs up during the AMD analysis or you don't have any
audio coming to asterisk through your sip channel.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Hawkin
Sent: April-23-09 11:00 AM
To:
I have the below script that doesn't seem to be working. I don't know if
I have something in the script wrong that I am just missing. Or if I
don't have the php.ini set correctly for emailing
This is the CLI output
-- Executing [4099xxx...@port3_real:1] Goto(DAHDI/50-1, newhire,s,1)
in
Check you can run the script from th ecommand line and successfully send
email... have you considered using phpagi for your scripts?
2009/4/23 James A. Shigley j...@answeringserv.com
I have the below script that doesn’t seem to be working. I don’t know if
I have something in the script wrong
Jimmy wrote:
Second Call out the asterisk console looks like
this-:
-- Executing [92952...@internal:1] Dial(SIP/222-09ab3588,
SIP/Cisco1760/2952210) in new stack
-- Called Cisco1760/2952210
[Apr 22 16:08:58] NOTICE[3450]:
you can try enabling agi debug on your console, you might be able to see
if there's an error on your agi script.
nhadie
James A. Shigley wrote:
I have the below script that doesn’t seem to be working. I don’t know if
I have something in the script wrong that I am just missing. Or if I
Martin wrote:
pri debug span 1
should show you the ISDN messages for 2BCT if there are any
Also someone should have told you that the 2BCT code is by default not
compiling
and you could enable it by editing chan_dahdi.c and adding
#define PRI_2BCT
Also since this flag is not
First run
/var/lib/asterisk/agi-bin/newhire.php
From linux command line to see if you don't have any error and that your AGI
is executable.
Then run 'agi debug' from the asterisk cli, place a call and see what was
send and receive from your agi
From:
Jared Smith wrote:
On Wed, 2009-04-15 at 09:58 -0500, Kevin P. Fleming wrote:
It's not enabled by default because when it is used the Asterisk server
loses control of the call and the CDR becomes incomplete. Not everyone
wants that behavior.
But since many people *would* like that behavior,
Don Kelly wrote:
Someone referred to a facility message when the TBCT call is torn down.
There are actually two messages--when the PSTN switch takes back the calls
and completes the transfer, it sends a facility message including a unique
ID. Then, when one of the parties disconnects, the
Max Metral wrote:
Ok, so I’ve made progress on 2BCT (2 B-Channel Transfer). I’m assuming
that the debug info below shows that XO doesn’t have 2BCT enabled on my
line, but if anybody can confirm that’ll let me be way more indignant. J
It would appear that the switch doesn't like what you're
Hello! Ive an issue whit CDR using asterisk 1.4.23.1. Ive configured mysql
to store cdr information, but, while I put into cdr_mysql.conf the field
userfield=1 and doing a query I found that this field is empty in the cdr
table. On the other hand I cant find records in the cdr table that show
This is the right suggestion:
Run something like the following:
[h...@mouth tmp]# echo this is a test | newhire.php
If the script runs, check your maillog (/var/log/maillog) to see if
there's any evidence of what may have happened.
Geraint Lee wrote:
Check you can run the script from th
Rob Hillis wrote:
Kurian Thayil wrote:
On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote:
Daily Asterisk restart
Do you think its mandatory in production env?
Daily? No. However, after implementing a weekly restart of Asterisk,
I've found the instance of lockups
On Thu, 23 Apr 2009, Darrick Hartman wrote:
Rob Hillis wrote:
Kurian Thayil wrote:
On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote:
Daily Asterisk restart
Do you think its mandatory in production env?
Daily? No. However, after implementing a weekly restart of Asterisk,
Actually I feel like an idiot. I had forgotten to put asterisk as an
allowed sender in the server that those emails are going out of.
(different from what * normally uses to email us)
James Shigley
Monroe Telephone Answering Service
409-981-9213
Infinity 5.5,UC 4.02.3803, Blink 3.0.104
The Asterisk.org development team has announced the release of Libpri
version 1.4.10. This release contains bug fixes related to handling of
BRI PTMP links and how messages are handled during link state
transients, as well as other fixes. Please see the Changelog for more
details.
The
Hello list.
I posted this over on the Biz section but some of the members thought
I might find more people running Asterisk on the Mac over here.
Here's my question:
I have looked at PHLink and PhoneValet and neither seem to be able to
do what I need, so I am looking at Asterisk.
What I want
Not true for me, a restart, if not mandatory, is a good idea, let's say
weekly or more depending on the usage.
My side is asterisk servers as gateways with thousands of calls a day
(at least 10.000 minutes a day).
I have asterisk(s) running for a long time( 4 years and more), with ss7
links
Un-top-posting and de-crufting, with some comments cast inline...
2009/4/23 James A. Shigley j...@answeringserv.com
mailto:j...@answeringserv.com
I have the below script that doesn?t seem to be working. I don?t
know if I have something in the script wrong that I am just missing.
Dan thank you, yes that seems to help. It looks like the bridging is happening
now and I see the light come on in the second FXO port, but then I get a busy
signal after that and the call still does not complete. If I set the second
line as priority 1 it completes the first call on that line
Gustavo A Gonzalez escribió:
Hello! I've an issue whit CDR using asterisk 1.4.23.1. I've configured
mysql to store cdr information, but, while I put into cdr_mysql.conf
the field 'userfield=1' and doing a query I found that this field is
empty in the cdr table. On the other hand I can't find
On Thu, 23 Apr 2009, Rick Dwyer wrote:
What I want to do is allow callers to call a our phone line and
unsubscribe their phone number from our call center list. So,
basically, when they call in, they would be greeted with a message
something like: please enter your 10 digit phone number
On Apr 23, 2009, at 3:14 PM, Rick Dwyer wrote:
Hello list.
I posted this over on the Biz section but some of the members thought
I might find more people running Asterisk on the Mac over here.
Here's my question:
I have looked at PHLink and PhoneValet and neither seem to be able to
do
On Apr 23, 2009, at 3:34 PM, Steve Edwards wrote:
On Thu, 23 Apr 2009, Rick Dwyer wrote:
What I want to do is allow callers to call a our phone line and
unsubscribe their phone number from our call center list. So,
basically, when they call in, they would be greeted with a message
On Apr 23, 2009, at 3:40 PM, Niles Ingalls wrote:
On Apr 23, 2009, at 3:14 PM, Rick Dwyer wrote:
Hello list.
I posted this over on the Biz section but some of the members thought
I might find more people running Asterisk on the Mac over here.
Here's my question:
I have looked at
My .02 - you should verify the number and ask for a (numeric) password.
This will save a good deal of grief.
Any interest in looking up the number in the database so they know
if they
are entering a subscribed number? How will you keep a disgruntled
customer
or employee from
How about posting a list of requirements on the biz list and soliciting
bids?
Eric Fort
FortConsulting
On Thu, Apr 23, 2009 at 12:47 PM, Rick Dwyer rdw...@quick-link.com wrote:
I don't know a thing about Linux and even on the Mac, my command line
skills are basic. So I would really be
Hi,
I've tried the link
http://www.asteriskguru.com/tools/audio_conversion.php but it returns an error
at the moment.
Any other ideas most welcome.
Tx Shaun___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
On Apr 23, 2009, at 3:58 PM, Eric Fort wrote:
How about posting a list of requirements on the biz list and
soliciting bids?
Good advice... I will probably do so tomorrow after I talk to
Ovolab... they have a product for OS X called Phlink they say can do
what I need.
Thanks,
--Rick
Somebody knows if I can save files in mp3 with the Record command on Asterisk?
I try to recompile sox to suport mp3 but Asterisk return the folowing message
when I use the Record command:
- Executing [...@liberado15:15] Record(SIP/1201-083453c8,
Hi,
My extensions.ael file includes :
context mylocal {
7530 = {
Dial(SIP/7530,,${OPTION});
NoOp(Here1);
};
7531 = {
Dial(SIP/7531);
NoOp(Here2);
};
};
If extension 7530 receives a call and transfer
The way I read to do this is to use sox to create a wav file, then use lame
to convert the wav to mp3. I did this for some MOH files.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jose Enes
Mateus
Sent: Thursday, April
On Thu, 23 Apr 2009, Rick Dwyer wrote:
I don't know a thing about Linux and even on the Mac, my command line
skills are basic. So I would really be looking for a GUI to configure
it, regardless of platform. I assume this is available on Linux?
There are web based interfaces available that
On Thu, Apr 23, 2009 at 10:08:39PM +0200, Shaun Wingrin wrote:
Hi,
I've tried the link
http://www.asteriskguru.com/tools/audio_conversion.php but it returns an
error at the moment.
sweetmorn*CLI help file convert
Usage: file convert file_in file_out
Convert from file_in to file_out.
Anyone know where I could find a good beginning for using asterisk and the
text based game adventure together such that I could play from the nearest
phone?
Thanks,
Eric
___
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Thanks!Ive solve the issue setting: unanswered=yes on cdr.conf .
Cheers!
Gustavo A. González
Dto. de Infraestructura
Despegar.com, Inc.
ggonza...@despegar.com
___
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You would just use festival/Cepstral combined with Read and an AGI to take
options and speak the result back. As long as you had a reasonably finite
number of possible outcomes, you could even do this just from a dialplan
without AGI.
_
From: asterisk-users-boun...@lists.digium.com
Hi,
When a SIP hardphone is transfering a call while ringing (caller and callee
don't speak to each other) using phone's Transfer key, it seems
BLINDTRANSFER remains empty.
Though I can see a 302 MOVED TEMPORARILY message coming in.
Is there a work around or something obvious I'm missing (it's
A good place to start is here:
http://www.venturevoip.com/news.php?rssid=1513
FreePBX includes a module called 'Zoip' which allows you to play Zork via a
Text-to-speech engine.
Why on Earth someone would want to do so is beyond me but hey... why not. :-)
Tim Nelson
Systems/Network
Eric Fort wrote:
Anyone know where I could find a good beginning for using asterisk and
the text based game adventure together such that I could play from
the nearest phone?
google on collossal cave.
honestly its the absolute worst unreadable mess of code ever conceived
by man or beast.
We are in a similar situation to you as far as moving from Cisco to
Asterisk. I have not got to the point of integrating Asterisk
directly to our PSTN gateways yet, but this might help.
On our H.323 gateways we use trunk groups for outbound call hunting.
You can create a single trunk group for
On Thu, 23 Apr 2009, Eric Fort wrote:
Anyone know where I could find a good beginning for using asterisk and the
text based game adventure together such that I could play from the nearest
phone?
Heh... I wrote a MUD once (still online, but..)
Key 1 to get the long sword.
Key 2 to go north.
hi all;
I need to find out about how to configure Asterisk (h323.conf) and
Audiocodes FXO/H323 voip-gateway.Audiocodes side too complex for that.
thank you so much
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
On 24/04/2009 1:11 a.m., Geraint Lee wrote:
Thanks for that, it's pretty much confirming what i first anticipated...
my intentions are as follows:
agents register with opensips, opensips clusters a set of call recording
servers which then connect to our border servers which will save cdr and
On 24/04/2009 3:00 a.m., Sam Hawkin wrote:
Hi All,
I am trying to use the AMD (Answering Machine Detect).
But it is not sending the AMD_Status as either
the Human or Machine, it hangs up in middle.
I'd say that the remote end of the call is hanging up - do a SIP debug
so you can see what
On 24/04/2009 2:24 a.m., Nhadie wrote:
Hi,
i'm currently using Originate command on AMI, i can call a certain
channel like a SIP user SIP/1000 then once 1000 is answered it dials out
to amobile or landline.
Would just like to know if i can use AMI to dialout to a mobile or
landline first
On Thu, Apr 23, 2009 at 6:11 PM, Matt Riddell li...@venturevoip.com wrote:
On 24/04/2009 2:24 a.m., Nhadie wrote:
Hi,
i'm currently using Originate command on AMI, i can call a certain
channel like a SIP user SIP/1000 then once 1000 is answered it dials out
to amobile or landline.
On 24/04/2009 10:19 a.m., Steve Totaro wrote:
A much more scalable way to do this is to create and then FTP or move
.call files to the proper directory. Depends how much you plan on
banging on the AMI.
Maybe, but the Asterisk Manager is happy with 10 calls per second and if
your controlling
On Thu, Apr 23, 2009 at 6:43 PM, Matt Riddell li...@venturevoip.com wrote:
On 24/04/2009 10:19 a.m., Steve Totaro wrote:
A much more scalable way to do this is to create and then FTP or move
.call files to the proper directory. Depends how much you plan on
banging on the AMI.
Maybe, but
Olivier wrote:
When a SIP hardphone is transfering a call while ringing (caller and
callee don't speak to each other) using phone's Transfer key, it seems
BLINDTRANSFER remains empty.
Though I can see a 302 MOVED TEMPORARILY message coming in.
If the person performing the transfer has dialed
Darrick Hartman wrote:
Rob Hillis wrote:
Daily? No. However, after implementing a weekly restart of Asterisk,
I've found the instance of lockups and CPU utilisation spikes have
decreased significantly.
Unless you're using some unstable modules, there really should be no
need to
Rob Hillis wrote:
Darrick Hartman wrote:
Rob Hillis wrote:
Daily? No. However, after implementing a weekly restart of Asterisk,
I've found the instance of lockups and CPU utilisation spikes have
decreased significantly.
Unless you're using some unstable modules, there really
On Thu, Apr 23, 2009 at 6:12 PM, Matt Riddell li...@venturevoip.com wrote:
On 24/04/2009 3:00 a.m., Sam Hawkin wrote:
Hi All,
I am trying to use the AMD (Answering Machine Detect).
But it is not sending the AMD_Status as either
the Human or Machine, it hangs up in middle.
I'd say
Hi all,
I'm trying to compile dahdi-tools on Ubuntu 8.04 on Xen (Amazon EC2 to be
exact). dahdi-linux compiled and installed successfully, after which I do the
following to install dahdi-tools:
wget
http://downloads.digium.com/pub/telephony/dahdi-tools/dahdi-tools-current.tar.gz
tar xzvf
On Fri, Apr 24, 2009 at 09:54:15AM +1000, i...@ameri.me wrote:
Hi all,
I'm trying to compile dahdi-tools on Ubuntu 8.04 on Xen (Amazon EC2 to be
exact). dahdi-linux compiled and installed successfully, after which I do the
following to install dahdi-tools:
wget
Darrick Hartman wrote:
If I were to do things again, I'd be running Astlinux on a net 5501 with
an integrated hard drive (for voicemail/IVR and so on) Only time I've
ever had to reboot my Astlinux box at home (on an ALIX-3) is when it's
time to upgrade Astlinux.
That's what we like to
On Fri Apr 24 2009 10:33:28 GMT+1000 (EST) Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
Everything seems to go well, the installation is successful, and I can start
the dahdi service after this and test it and it all seems fine.
The issue then is that my whole system becomes unusable. It
I have written an asterisk manager client which creates an outbound call
using Asterisk manager API's Originate action.
when the call is connected I run 3 applications on it.
1)read a dtmf digit from user
2)A customized application which I have written,(It plays something to user)
3)Hangup
If
hi i need many cheaps atas or some very cheap way to connect analogs phones
to asterisk
what do you recomend? i searches and only find solutions like 40 U$D (in the
states, here in argentina is like 80 U$D) per phone any links or something?
thanks!
David
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(='.'=)This is Bunny. Copy and
On Fri Apr 24 2009 14:03:14 GMT+1000 (EST) David fire ddf...@gmail.com wrote:
hi i need many cheaps atas or some very cheap way to connect analogs
phones to asterisk
what do you recomend? i searches and only find solutions like 40 U$D (in
the states, here in argentina is like 80 U$D) per
thanks for your answer but the boards are too expensive.
4 fxs digium card is like 600.
the 12 fxs open vox board (i have one) is like 40U$D per phone. (in the
states).
i need cheap!!!
David
2009/4/24 Aryan Ameri i...@ameri.me
On Fri Apr 24 2009 14:03:14 GMT+1000 (EST) David fire
2009/4/24 Kevin P. Fleming kpflem...@digium.com
Olivier wrote:
When a SIP hardphone is transfering a call while ringing (caller and
callee don't speak to each other) using phone's Transfer key, it seems
BLINDTRANSFER remains empty.
Though I can see a 302 MOVED TEMPORARILY message coming
Have you checked ebay?
David fire wrote:
hi i need many cheaps atas or some very cheap way to connect analogs
phones to asterisk
what do you recomend? i searches and only find solutions like 40 U$D
(in the states, here in argentina is like 80 U$D) per phone any links
or something?
Jimmy wrote:
Dan thank you, yes that seems to help. It looks like the
bridging is happening now and I see the light come on in
the second FXO port, but then I get a busy signal after
that and the call still does not complete. If I set the
second line as priority 1 it completes the first
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