On Mon, 2010-05-24 at 11:51 -0500, Danny Nicholas wrote:
The Cepstral paid version has several languages available and other voices
for those 10 people who don't like Allison. At $35.00 a pop, it's not
prohibitive (Lumenvox is much more pricey)
This is the initial cost, there are also per
I am testing Jack in asterisk 1.6.2.7 now, using JACK_HOOK and channel
variables (connecting parties to jack during the call).
It works, but it chokes. Every 2-3 days hangs my asterisk. Debugging
it now, trying to find the reason.
So far could run only the standard Debian repo jack (0.109.2,
Hi!
I tried with 0.109.* up to 0.112 or so. I once managed an ISDN call. but
with all newer versions googletalk somehow failed. There is a bugreport in the
system, but unfortunitely it still seems unresolved:
http://bugs.digium.com/view.php?id=13812
Now I'm running Jack 0.116.2, which
Hello List,
I think I've discovered a little bug in t.38 bug in 1.6.0.22
regarding the speed (T38MaxBitRate) used to send the faxes.
Asterisk always responds with a=T38MaxBitRate:2400. I've tried
with Patton and Grandstream devices and the result is always the
You are completely right!
You are my hero!
I'm experiencing the same error with hylafax + t38modem implementation, and
t38modem ALLWAYS sends out at 2400!!!
I have 1.6.0.22 too, so this is definetly what's happening!
1 month looking for the error and now Alexandru has it...whoa!
regards,
Miguel
On 05/25/2010 05:48 AM, Alexandru Oniciuc wrote:
Hello List,
I think I’ve discovered a little bug in t.38 bug in
1.6.0.22 regarding the speed (T38MaxBitRate) used to send the faxes.
Asterisk always responds with a=T38MaxBitRate:2400.
I’ve tried
Hi Kevin,
Thanks for your quick reply.
I have been trying to find a good configuration for T38modem + hylafax +
asterisk for almost 1 month, and the issue is allways the same: just syncs
at 2400 bpps.
Help on this issue would be apreciated.
Regards,
Miguel Amez
2010/5/25 Kevin P. Fleming
On 05/25/2010 07:54 PM, Kevin P. Fleming wrote:
On 05/25/2010 05:48 AM, Alexandru Oniciuc wrote:
Hello List,
I think I’ve discovered a little bug in t.38 bug in
1.6.0.22 regarding the speed (T38MaxBitRate) used to send the faxes.
Asterisk always
Hi all,
I have asterisk 1.4.26 (and I tried 1.4.29) connected PRI all 23 lines
and for the
most part everything works. Dialing out on 23 lines to phones works fine.
I have to use the Local channel to call the intercom system (from call
files).
If I only call 1 intercom system at a time so it
On Tuesday 25 May 2010 08:28:57 Steve Underwood wrote:
On 05/25/2010 07:54 PM, Kevin P. Fleming wrote:
On 05/25/2010 05:48 AM, Alexandru Oniciuc wrote:
if ((sscanf(a, T38FaxMaxBuffer:%30u,x) == 1)) {
ast_debug(3, MaxBufferSize:%d\n, x);
found = TRUE;
Jerry Geis wrote:
Hi all,
I have asterisk 1.4.26 (and I tried 1.4.29) connected PRI all 23 lines
and for the
most part everything works. Dialing out on 23 lines to phones works fine.
I have to use the Local channel to call the intercom system (from call
files).
If I only call 1
I found I was not setting the TOUCH_MONITOR variable properly!
Thank you to all.
Regards,
Marta
On 18 May 2010 09:17, Marta Silva mrtsil...@gmail.com wrote:
Hi there,
We used to record all the calls with the Monitor function.
Now, I haveimplemented on-demand recording with automon
Running 1.6.2.8 and using teliax and junction for sip providers. We also
have an internal sip network for extensions. On boot both providers
are shown as unspecified and UNKNOWN. The internal sip extensions are
found. sip reload always finds the providers.
Why doesn't asterisk find the
Does anyone know what commands in the config file for a SuSE Firewall will
forward 5060 and RTP ranges to an Asterisk box in the internal LAN?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
I have tried chan_h323 and chan_ooh323 but have not been abled to get video
between endpoints.
I was wondering if there was a way to enable video or if it is even supported
at this point?
--
_
-- Bandwidth and Colocation
Hi,
I am having very strange situation. I have my sip peer located over the
internet and I am able to connect and dial to it.
the problem is that, if yesterday i was connected with it via my asterisk
client and dialing to it normal way, today when i run asterisk on my client,
my sip peer
Brent A. Torrenga wrote:
Does anyone know what commands in the config file for a SuSE Firewall
will forward 5060 and RTP ranges to an Asterisk box in the internal LAN?
I think you need to play with the parameter FW_FORWARD_MASQ in the
/etc/sysconfig/SuSEfirewall2
For example:
By browsing on the mailing list I learned that its possible to generate .h263
asterisk friendly files with gstreamer.
The script below it's supposed to do just that, however I get error when trying
it out locally.
gst-launch filesrc location=AstriDevCon_Europe_2006.mov ! qtdemux name=demux !
I want to set up a conference call to be recorded automatically, so
I'd like the recording to start when the second caller joins the
conference (one caller already there). The recording would continue
until the last user hangs up.
How can you determine how many are already in the conference
On Tue, 2010-05-25 at 12:07 -0600, Steve Johnson wrote:
How can you determine how many are already in the conference bridge?
I don't know that there's a way to do it automagically within
ConfBridge. I use the GROUP() and GROUP_COUNT() functions to do these
sorts of things.
--
Jared Smith
The Asterisk Development Team has announced the release of versions
2.3.0.1 and 2.2.1.2 of DAHDI Linux. These releases are available for
immediate download at
http://downloads.asterisk.org/pub/telephony/dahdi-linux.
Both these releases address system compatibility issues with the
wcte12xp and
I work at home with standard residential cable Internet service and I wanted to
test CPA for use with our dialer solution. The first problem I ran into is that
CPA only works with a SIP provider that does IP based authentication opposed to
usename/password authentication. After I got an account
I can run 1.6.0 with at least twice as many calls as 1.6.1 and 1.6.2
with the same hardware. Once I get about 180 calls on 1.6.1 or 1.6.2
they max out the cpu and crash.
Try 1.6.0.28 and see if that works better would be my suggestion.
John
-Original Message-
On Mon, May 24, 2010
I was at a client site tonight to install OSLEC on his machine running
asterisk 1.6.0.22 and DAHDI 2.2.1 installed via yum. I stopped asterisk and
DAHDI, downloaded the latest version of DAHDI 2.2.1
(dahdi-linux-complete-2.2.1.2+2.2.1.1) and made the necessary changes to
compile OSLEC with DAHDI,
Hi Everybody,
Im getting as strangeous issue on Asterisk 1.4.31 (Using Elastix) ...
In some calls, i get an atxfer or musiconhold in the middle of call, or
listening another call (like a cross line) without any intervention of
the user. I got this error in about 3-10% of the calls, on a
Are your extensions(who get the music between the calls) on SIP ?
When the issue occurs, note
1. the SIP peer account with which it is occurring
2. Without hanging up, do a core show channels to see how many channels
are present for that same SIP peer. If your are unable to identify this
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