Hellos,
I am running asterisk 1.2 with linksys spa 942. Normally, when a call calls
an extension and the extension is on another call, the call is put in line
but not notified that the extension is busy. I want to notify the second,
third and fourth callet that they are on extension is on another
Hi,
I need to replace digital handsets in offices where there cabling is
appareantly not Ethernet-compliant.
Today's usage is to press a key to toggle between private ou public line
before issuing an outgoing call.
Are you aware of a DECT handset (to overcome cabling limitations) that mimic
this
2009/8/18 Olivier oza-4...@myamail.com
Hi,
I need to replace digital handsets in offices where there cabling is
appareantly not Ethernet-compliant.
Today's usage is to press a key to toggle between private ou public line
before issuing an outgoing call.
Are you aware of a DECT handset (to
Would it be possible to execute some kind of script when for example
Asterisk restarts... or stops... ?
How can one read the status of Asterisk so that when the service is
stopped I could be notified by mail, by text message,... ?
I don't know how to read the status of Asterisk (or the change of
Hi,
maybe you wan't to use '0' in front of you telephone number. eg.
intern: 261 - 261
exten: 002151-5462 - 021515462
Bye
On Tue, Aug 18, 2009 at 9:08 AM, Olivieroza-4...@myamail.com wrote:
Hi,
I need to replace digital handsets in offices where there cabling is
appareantly not
You can also use different identities.
On Tue, Aug 18, 2009 at 9:08 AM, Olivieroza-4...@myamail.com wrote:
Hi,
I need to replace digital handsets in offices where there cabling is
appareantly not Ethernet-compliant.
Today's usage is to press a key to toggle between private ou public line
On 18/08/09 08:08, Olivier wrote:
Hi,
I need to replace digital handsets in offices where there cabling is
appareantly not Ethernet-compliant.
Today's usage is to press a key to toggle between private ou public line
before issuing an outgoing call.
Are you aware of a DECT handset (to
Hi,
well there are differnt ways to do it. It depends on what you want.
The start-stop scripts in /etc/init.d/ are looking for a pid file, so
they can figure out if the server is running. You can change the
script to get a message if the server is going up or down by the
script.
If you want that
On Mon, 17 Aug 2009, Pascal Bruno wrote:
Not sure if anybody noticed, but it seems like Skype For Asterisk is out.
$66 per channels, pretty pricey
http://store.digium.com/productview.php?product_code=1SFA0001
Yes, pretty pricey indeed especially considering that you can buy Skype
ATA
Hi
Is it any interest to use realtime applications if I use mysql to store my
sip terminal informations ?
Can I use some informations from mysql databese and sip.conf in same time ?
Anyone already use Asterisk with Active Directory for centralized database
for authentication information and
On Tuesday, August 18, 2009, Remco Barendse wrote:
But then again, who needs Skype for business purposes anyways, i
don't think there is a huge market for it.
Me ... at least in theory! Our cellphones have built-in Skype, so a
Skype gateway should give me call forwarding and diversion to our
hi,
you can use call-limit=1 in sip.conf or DEVSTATE()
http://www.voip-info.org/wiki/view/Asterisk+func+device_State
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
http://www.voip-info.org/wiki/view/Asterisk+sip+incominglimit
bye
On Tue, Aug 18, 2009 at 9:03 AM, James
On Tue, 18 Aug 2009, Geoff Lane wrote:
On Tuesday, August 18, 2009, Remco Barendse wrote:
But then again, who needs Skype for business purposes anyways, i
don't think there is a huge market for it.
Me ... at least in theory! Our cellphones have built-in Skype, so a
Skype gateway should
On Tue, 18 Aug 2009, Olivier wrote:
Hi,
I need to replace digital handsets in offices where there cabling is
appareantly not Ethernet-compliant.
Today's usage is to press a key to toggle between private ou public line
before issuing an outgoing call.
Are you aware of a DECT handset (to
On Tuesday, August 18, 2009, Gordon Henderson wrote:
I was under the impression that Three (who I guess you're using)
placed a regular call over their network then Skyped it at their
HQ - rather than have the Skype client actually reside in the
handset.. (And I'm suspecting their 3G
Good luck with the N95... my experiences of the N95 and SIP haven't been
great... the phone likes to restart... regularly. Nokia may well have fixed
these glitches by now though. Getting it configured was a bit of a mission
too... and as expected the battery life shoots down when it's enabled...
hi,
stunaddr = stun.exiga.net looks wrong ^^
in generally it looks like a nat problem.
bye,
patrick
On Mon, Aug 17, 2009 at 8:12 PM, Daniel Bareirodaniel-lis...@gmx.net wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I'm trying to connect to ekiga.net through a client
On Tue, 18 Aug 2009, Geraint Lee wrote:
Good luck with the N95... my experiences of the N95 and SIP haven't been
great... the phone likes to restart... regularly. Nokia may well have fixed
these glitches by now though. Getting it configured was a bit of a mission
too... and as expected the
Thanks for the reply Patrick. I do not want to limit the second, third...etc
call, but I want the caller to be notified that the extension they are
calling is on another call and they are waiting. Let me look into DEVSTATE
On Tue, Aug 18, 2009 at 12:12 PM, Patrick Plattes
In fact, I am not using Asterisk-java.
I use asterisk and fastagi to have a b2bua able to disconnect many
callers if their customer account is around 0.
Is that a bug to be reported?
How to solve that in Java without using Asterisk-java?
Kind regards,
Olivier
Stefan Reuter a crit:
Yes
Modify your code to support 1.6-style events.
hh174 wrote:
In fact, I am not using Asterisk-java.
I use asterisk and fastagi to have a b2bua able to disconnect many
callers if their customer account is around 0.
Is that a bug to be reported?
How to solve that in Java without using
Of course it has been done, but I don't think it's a clean way?
Having to send a fake command to asterisk to be able to read the
'hangup' isn't really clean.
I must suppose there is a clean solution for that.
Kind regards,
Olivier
Alex Balashov a écrit :
Modify your code to support 1.6-style
Geoff Lane wrote:
On Tuesday, August 18, 2009, Gordon Henderson wrote:
I was under the impression that Three (who I guess you're using)
placed a regular call over their network then Skyped it at their
HQ - rather than have the Skype client actually reside in the
handset.. (And I'm
2009/8/18 Patrick Plattes patr...@erdbeere.net
You can also use different identities.
Yes, it's true but the trouble is to find a convenient way to switch from
identity to another ...
At the moment, user would press Identity1 key, then dial.
___
--
2009/8/18 Alan Lord (News) alansli...@gmail.com
On 18/08/09 08:08, Olivier wrote:
Hi,
I need to replace digital handsets in offices where there cabling is
appareantly not Ethernet-compliant.
Today's usage is to press a key to toggle between private ou public line
before issuing an
Hi,
what's about the Snom m3? You can use 8 different identities. There
should be a documentation on the Website how to change the identities.
I don't have any experience with the m3s, so you should take care of
the stupidly long-winded sequence sequence to switch accounts, as
gordon wrote.
You
I want to install Dahdi and Dahdi-tools on a CentOS 5.3 Xen host and I
receive the following error :
You do not appear to have the sources for the 2.6.20-prep kernel
installed.
I have installed :
- kernel-headers-2.6.18-128.4.1.el5.x86_64
- kernel-devel-2.6.18-128.4.1.el5.x86_64
-
On Tue, Aug 18, 2009 at 02:07:39PM +0200, jonas kellens wrote:
I want to install Dahdi and Dahdi-tools on a CentOS 5.3 Xen host and I
receive the following error :
You do not appear to have the sources for the 2.6.20-prep kernel
installed.
I have installed :
-
On 19/08/09 12:07 AM, jonas kellens wrote:
I want to install Dahdi and Dahdi-tools on a CentOS 5.3 Xen host and I
receive the following error :
You do not appear to have the sources for the 2.6.20-prep kernel
installed.
I have installed :
- kernel-headers-2.6.18-128.4.1.el5.x86_64
-
Hi,
I'm trying to achieve the following feature that's common in Avaya
systems:
A user page all extensions in a full duplex mode- he can hear all, and
all can hear him via their phones' speaker. When one of the extensions
picks up the handset, the call is bridged between the pager and the
Hello
I have two questions !
1. What is the best speech recognition engine for asterisk? I have searched
and asked on forums and found that lumen vox is best for asterisk bala bla
bla
2. For TTS (text to speech) which TTS engine will be better to use ? I have
tested Flite , cepstral (i have not
ABBAS SHAKEEL wrote:
1. What is the best speech recognition engine for asterisk? I have
searched and asked on forums and found that lumen vox is best for
asterisk bala bla bla
Well, if you're going to ask a question to which the answer could,
conceivably, be the one you are suggesting
On Tue, 18 Aug 2009 17:29:27 +0500, ABBAS SHAKEEL wrote:
Take a look to loquendo !
Hello
I have two questions !
1. What is the best speech recognition engine for asterisk? I have
searched and asked on forums and found that lumen vox is best for asterisk
bala bla bla
2. For TTS
On Tue, 18 Aug 2009, Patrick Plattes wrote:
Hi,
what's about the Snom m3? You can use 8 different identities. There
should be a documentation on the Website how to change the identities.
I don't have any experience with the m3s, so you should take care of
the stupidly long-winded sequence
Thanks for your reply.
I am renting a Virtual Dedicated Server and installing Asterisk on it. I
do not have much to say about the Xen-image that is used by my Hosting
Provider...
But I do have root-access and am able to do everything I want :-).
The image that is used is : centos53-bare.conf
Hello
I have upgraded our asterisk box from zaptel to dhadi two weeks ago...
Since, there has been quite a significant amount of echo when making a
call. Only for the local outgoing call, the person on the other side
doesn't hear any echo.
This is with a TE-110P ISDN PRI card ..
I've pretty
Jean-Yves Avenard wrote:
From reading the various documentation, I was convinced that moving
from zaptel to dahdi was almost just a matter of renaming the
configuration file... Am I mistaken ?
Did you read the upgrade documentation that comes with DAHDI,
specifically from UPGRADE.txt:
* It
On Tue, Aug 18, 2009 at 10:50:25PM +1000, Jean-Yves Avenard wrote:
Hello
I have upgraded our asterisk box from zaptel to dhadi two weeks ago...
Since, there has been quite a significant amount of echo when making a
call. Only for the local outgoing call, the person on the other side
Hi
That was a fast answer, impressive !
2009/8/18 Kevin P. Fleming kpflem...@digium.com:
Did you read the upgrade documentation that comes with DAHDI,
specifically from UPGRADE.txt:
I did, but I guess I did not pay enough attention...
* It is no longer possible to select a software echo
On Tue, 18 Aug 2009, Kevin P. Fleming wrote:
[snip]
Note: It is *mandatory* to configure an echo canceler for the
system's channels using dahdi_cfg unless the interface cards in use
have echo canceler modules available and enabled. There is *no*
default software echo canceler with
Hi
2009/8/18 Tzafrir Cohen tzafrir.co...@xorcom.com:
Something is missing here...
http://docs.tzafrir.org.il/dahdi-tools/#_echo_canceller_modules
Thanks ..
I added to /etc/dahdi/system.conf the following:
echocanceller=mg2,1-10
However, I have no clue about the various echo canceller,
On Tue, 18 Aug 2009, jonas kellens wrote:
Would it be possible to execute some kind of script when for example
Asterisk restarts... or stops... ?
How can one read the status of Asterisk so that when the service is
stopped I could be notified by mail, by text message,... ?
I don't know how
hi,
try something like this:
bchan=4-5
hardhdlc=6
echocanceller=mg2,4-5
and then it should look like:
Asterisk-1:~# dahdi_cfg -v
DAHDI Tools Version - SVN-trunk-r6902
DAHDI Version: SVN-trunk-r6946
Echo Canceller(s): MG2
Configuration
==
SPAN 1: CCS/ AMI Build-out: 0 db
Hello,
I've a problem. I've asterisk 1.6.0.5 version. And I've created
callcenter, but agents registers to another SIP server. When agent tries
transfer a client to another operator , pressing flash, I get this:
[Aug 18 16:06:37] WARNING[5259]: chan_sip.c:5349 sip_indicate: Don't know
how to
Jeff LaCoursiere wrote:
On Tue, 18 Aug 2009, Kevin P. Fleming wrote:
[snip]
Note: It is *mandatory* to configure an echo canceler for the
system's channels using dahdi_cfg unless the interface cards in use
have echo canceler modules available and enabled. There is *no*
default
Hello List,
our setup:
Callcenter
IBM Hardware, 1x TE420, 1x xircom analog switch, 4x different cellular
providers on the xircom analog port, ~60 agents
Debian 5.0.1 (Lenny)
Asterisk 1.4.21.2 Debian Package recompiled with additional app_queue
segfault fix
Zaptel 1.4.11 Debian Package
My
Rajkumar S escribió:
Hi,
I am using RT engine to log queue_log to a mysql database. My extconfig is
[settings]
queue_log = mysql,asterisk16_production
Logging to mysql is working fine.
But I find that the queue_log file now only has QUEUESTART lines for eg:
Kevin P. Fleming wrote:
Jeff LaCoursiere wrote:
On Tue, 18 Aug 2009, Kevin P. Fleming wrote:
[snip]
Note: It is *mandatory* to configure an echo canceler for the
system's channels using dahdi_cfg unless the interface cards in use
have echo canceler modules available and enabled.
Dave Fullerton wrote:
Here's my $0.02. If you don't want an echo canceller, specify
echocanceller=none,x-y and have dahdi_cfg print a warning (at any
verbosity level) when an echo canceller is not specified for a channel.
Personally, I would also like to see an option that says Use the
anyone already used the realtime driver for LDAP in order to interact
Astérisk with Active Directory ?
regards
Harry
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Register
I'm making outbound calls by placing call files in the asterisk outgoing
directory. At times, the call would be hung by SS7 without even
attempting (due to error in the outgoing number). I get the following on
console:
-- Attempting call on ss7/9297210213 for s...@croom:1 (Retry 1)
--
If you make your call files use a context, instead of a number, HANGUPCAUSE
will be available to you. For example, let's say your call file now does
this:
Channel: DAHDI/2/ww12054918802
CallerID: SIP/104
MaxRetries: 1
WaitTime: 45
retryTime: 5
Application: record
Data: /tmp/DADHI2.wav
This calls
I would have happily bought 20 channels at $10/channel, but at most will
be buying only a single channel now :\
Pascal Bruno wrote:
Not sure if anybody noticed, but it seems like Skype For Asterisk is out.
$66 per channels, pretty pricey
hello,
please any IAX2 ActiveX control that wrap libiax2 or libiaxclient?
i want to develope my softphone in delphi
thanks
__ Information from ESET NOD32 Antivirus, version of virus signature
database 4345 (20090818) __
The message was checked by ESET NOD32 Antivirus.
http
Pricing is a very legitimate way to minimise support effort. It winnows
down the market size to a point where the company offering the goods
can sustain the projected per user support issues.
You can always drop the price later on when you have a better handle on
the per user support issue.
Casey Boone wrote:
I would have happily bought 20 channels at $10/channel, but at most will
be buying only a single channel now :\
That does sound a bit pricey, although it it's as stable as the latest
beta, I wont be buying it at all.
___
--
I'm going blind searching - maybe you know?
During the execution of a script I want to play fake ring to caller.
Both of these examples complain of missing option:
$agi-exec(Ringing);
$agi-exec(Playtones ring);
Notice: Undefined variable: options in
Hello,
I have at my disposal, 2, TE410P cards. I have setup an asterisk box
that will function ONLY to handle meetme conferences over SIP. No PRI
lines to be plugged into the 410 card.
MeetMe requires an external timing source. Right now, using the dummy
driver. Is it possible to use the card
Thanks Alex Balashov , Sébastien Cramatte thanks
bla bla bla means etc etc..
Balashov you recomend lumenvox for speech recognition ?
Sebastien loquendo really looks interesting !
Can i get URDU language support ?
best regards
On Tue, Aug 18, 2009 at 5:41 PM, Sébastien Cramatte
Boehm, Matthew wrote:
MeetMe requires an external timing source. Right now, using the dummy
driver. Is it possible to use the card solely for timing purposes? Any
benefit to doing so? Or should I just sell the cards?
DAHDI 2.2.0 provides timing without using the dummy driver and without
On Tue, Aug 18, 2009 at 12:09 PM, Boehm, Matthewmbo...@theplanet.com wrote:
Hello,
I have at my disposal, 2, TE410P cards. I have setup an asterisk box that
will function ONLY to handle meetme conferences over SIP. No PRI lines to be
plugged into the 410 card.
MeetMe requires an external
Michael Graves wrote:
Pricing is a very legitimate way to minimise support effort. It winnows
down the market size to a point where the company offering the goods
can sustain the projected per user support issues.
You can always drop the price later on when you have a better handle on
the
Have you tried unloading and reloading the zaptel driver?
l.
2009/8/18 Raimund Sacherer r...@runsolutions.com
Hello List,
our setup:
Callcenter
IBM Hardware, 1x TE420, 1x xircom analog switch, 4x different cellular
providers on the xircom analog port, ~60 agents
Debian 5.0.1 (Lenny)
Lenz Emilitri escribió:
You should log to a file and use a piece of code like our qloaderd to
do the DB update.
l.
Could you share such piece of code?
Thanks,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
___
-- Bandwidth and
You should log to a file and use a piece of code like our qloaderd to do the
DB update.
l.
2009/8/17 Rajkumar S rajkum...@gmail.com
Hi,
I am using RT engine to log queue_log to a mysql database. My extconfig is
[settings]
queue_log = mysql,asterisk16_production
Logging to mysql is
Did you try use busytect option enabeled into zaptel.conf file ?
Another way must be recompile your asterisk an enable BUSYDETECT
options for hangup.
Regards,
On Wed, Aug 19, 2009 at 8:55 AM, Raimund Sachererr...@runsolutions.com wrote:
Hello List,
our setup:
Callcenter
IBM Hardware, 1x
Why not record a ring tone, and playback the file? with $agi-streamfile???
On Tue, Aug 18, 2009 at 11:38 AM, Barton Fisher bhfis...@icpage.com wrote:
I'm going blind searching - maybe you know?
During the execution of a script I want to play fake ring to caller. Both
of these examples
Lol but he has a good point and makes a lot of sense. Never thought about
that strategy...
On Tue, Aug 18, 2009 at 12:16 PM, Thomas Kenyon
dig...@sanguinarius.co.ukwrote:
Michael Graves wrote:
Pricing is a very legitimate way to minimise support effort. It winnows
down the market size to
Hello there!
During some research on Internet I found the following comparison on
site Voip-Info (see, http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ;):
The main points listed on Asterisk's CONS that concerned me were:
* Conferencing on Asterisk depends on Zaptel hardware and/or kernel
On Tue, Aug 18, 2009 at 1:52 PM, Mauro Sergio Ferreira Brasil
mauro.bra...@tqi.com.br wrote:
Hello there!
During some research on Internet I found the following comparison on
site Voip-Info (see, http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ
):
The main points listed on Asterisk's
Steve Totaro wrote:
On Tue, Aug 18, 2009 at 1:52 PM, Mauro Sergio Ferreira Brasil
mauro.bra...@tqi.com.br mailto:mauro.bra...@tqi.com.br wrote:
Hello there!
During some research on Internet I found the following comparison on
site Voip-Info (see,
It all depends what are you going to use Asterisk for. Sounds like it is
for conferencing. Would you care to elaborate?
CS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio
Ferreira Brasil
Sent:
Hi there,
I though to chime in here just to share my opinion for what is worth. As a
developer who enjoys playing with telephony in general I try to remain as
objective as possible when talking about one or the other, and I felt that
N from arcdiv was a bit unfair with FreeSWITCH docs.
Then
Moises Silva wrote:
Hi there,
I though to chime in here just to share my opinion for what is worth.
As a developer who enjoys playing with telephony in general I try to
remain as objective as possible when talking about one or the other,
and I felt that N from arcdiv was a bit unfair with
Could one use different family-names in the Asterisk Realtime
Architecture ??
Suppose I have 3 SIP-accounts, where incoming conversations al come into
a different [context].
Now I want to put the detail of the context into a mysql database.
So I was thinking :
[family] =
possibly very simple needs, but that doesn't make it very complete
documentation.
Which nobody has stated it is. Just means for your needs (which you have
never stated nor pointed to particular missing chunks of documentation for
what you intended to do) was far from fulfilling your
I just want to also remind people that Skype for SIP is also to be released
shortly. When I last talked to Skype they said it would be out in late
July. So I assume if you wait another few more weeks the entire issue will
be moot. No $60/channel fee, just the free SIP platform for people using
from ESET NOD32 Antivirus, version of virus
signature database 4345 (20090818) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
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On 19/08/09 3:25 AM, Meftah Tayeb wrote:
hello,
please any IAX2 ActiveX control that wrap libiax2 or libiaxclient?
i want to develope my softphone in delphi
http://www.geocities.com/babarnazmi/
--
Cheers,
Matt Riddell
Director
___
Man... I need to be very frank with you... I don't know any more.
We started analysing what can be done to get Asterisk working on a way
we want it to work, that is: totally dynamic dial plan generated by an
external server (responsible for business logic and legacy interface),
and retrieved
1250628579|MANAGER|S1140|Agent/265|ADDMEMBER|
Above is an example of a line from the file. Is there an option to
have this date/time output in a way that can be read without
conversion? The date/time in full and messages is formated so it can
be read directly.
--
Jim Dickenson
Hi everyone
Has anybody ever come across this, I want it to join automatically the
recording files for me but it creates the wav files as I expect but it
doesn't join them sadly, so eg:
ls
1250623104.10-in.wav 1250628672.23-in.wav 1250630411.30-in.wav
1250623104.10-out.wav
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dáibhéad Antoine O'Reilligh wrote:
Have I forgotten anything?
Do you have 'sox' installed on your asterisk box?
Barry
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (GNU/Linux)
Snipped from:
http://store.digium.com/productview.php?product_code=1SFA0001
Supports G.711 and G.729 (included) codec's
Do I understand correctly that my SFA-to-Skype calls would be throttled down
to 8khz even if not traversing the PSTN?
My biggest draw to SFA is 440+ million subscribers in a
As posted on blogs.digium.com today:
http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
the Asterisk project has changed providers for Music-On-Hold (MOH)
content distributed with/for Asterisk. In addition to the change for
future Asterisk releases, we have also opted to rebuild
I just solved this :)
Turns out that was removed in Asterisk 1.6 :) The solution is:
monitor-type = MixMonitor
monitor-format = wav
Hopefully will help somebody!! also has a really neat after-recording shell
command option, so after a file has been saved you can instantly have it
converted to
Guys
I need to send my extension number to a trunk, I'm using Asterisk
now, any idea how to do it?
Thanks
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To Members,
I currently looking for someone who can configure Cisco IAD's Ie.. IAD 2431
And yes we are more than willing to pay for the service.
If interested please drop me an email
m...@openaccessinc.commailto:m...@openaccessinc.com
Michael DiMartino | Director of IT | Open Access, Inc.
115
To Members,
I currently looking for someone who can configure Cisco IAD's Ie.. IAD 2431
And yes we are more than willing to pay for the service.
If interested please drop me an email
m...@openaccessinc.commailto:m...@openaccessinc.com
Michael DiMartino | Director of IT | Open Access, Inc.
115
Hi Lenz, thanks,
No, i did not try unloading / reloading the zaptel driver, would this
result in a loss of current calls?
As I stated our client has a callcenter and I can not risk loosing
calls (well, at times asterisk does this to me on itself with
segfaults/locks ...)
best
--
Karl Fife wrote:
To some degree, I (and I'm sure other wideband disciples), feel somewhat
like the only guy on the block with a fax machine. Per Metcalfe's law it
seemed that Skype (as a easily accessible namespace supporting wideband)
could have been a shot in the arm for 8khz telephony.
Hi Luis,
the problem is it is basically the first time i saw it (or recognized
it) so, it does definitly not happen regularly, I have more problems
with our xircom analog usb switch which handles our outgoing mobile
connections, this stuff has problems detecting busydetect but this I
On Tue, Aug 18, 2009 at 10:00:55AM -0400, Dave Fullerton wrote:
Kevin P. Fleming wrote:
Jeff LaCoursiere wrote:
On Tue, 18 Aug 2009, Kevin P. Fleming wrote:
[snip]
[snip]
Here's my $0.02. If you don't want an echo canceller, specify
echocanceller=none,x-y and have dahdi_cfg print
That does sound a bit pricey, although it it's as stable as the latest
beta, I wont be buying it at all.
Have you posted a bug describing the issues you are having at
http://betareports.digium.com/mantis/
yet? I would love to have the opportunity to actually fix any bugs
that people find.
- Original Message -
From: Kevin P. Fleming kpflem...@digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, August 18, 2009 5:03 PM
Subject: Re: [asterisk-users] Skype for Asterisk -- Codec support
Karl Fife wrote:
To
I just want to also remind people that Skype for SIP is also to be
released shortly. When I last talked to Skype they said it would be
out in late July. So I assume if you wait another few more weeks
the entire issue will be moot. No $60/channel fee, just the free
SIP platform for
Karl Fife wrote:
Any idea what timeframe?
Not that I can disclose, no. Sorry.
Can I assume that SFA licenses now will be valid for future releases?
Yes. It is quite unlikely (although not impossible, of course, like any
piece of software) that existing SFA licenses would not be valid for
Take a look:
1) Verify the cable pin out for rj-11 conector for analog por 6 and 9.
The pin out must be equal to any port that work fine.
2) If not (1) try to reproduce the scenary, call from iax client to
any celular. Activate the debug level and verbose level. (core set
debug 255, core set
Hello, all. I've solved my own problem but will post it here in case
someone else has the same misunderstanding in the future.
We thought we had set up our meetme so that regular users entered the
conference without a pin but could not speak to each other until the
moderator arrived. We
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
SIP wrote:
Daniel,
Hi SIP.
Check your stunaddr setting. Is it misspelled, or do they really use
stun.exiga.net instead of stun.ekiga.net ?
Thanks to indicate that error to me. I doing the test again. I don't
believe that this solves what I
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