Dear All
Can you please let me know if the asterisk has speech to text and text
to speech facilities?
Thank you
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We have Asterisk 1.6.2.17.2 connected to a sangoma E1 card. The problem
we are having is that we have a calling card type application and when
people enter the number to be dialled we call the Dial application. It
gets back an indication that the number is incomplete (via PRI cause
code 28 I
Hi
I'm using asterisk 1.8.3.2 (with a couple of patches)
I have the following scenario...
SIP call comes in and gets answered by extension A (MixMonitor is
executed as part of this inbound dial plan of the number being called)
Extension A puts call on hold and calls extension B
Extension A
Hi
I'm using asterisk 1.8.3.2 (with a couple of patches)
I have the following scenario...
SIP call comes in and gets answered by extension A (MixMonitor is
executed as part of this inbound dial plan of the number being called)
Extension A puts call on hold and calls extension B
On Tue, 2011-08-02 at 07:51 -0400, Dan Journo wrote:
Hi
I'm using asterisk 1.8.3.2 (with a couple of patches)
I have the following scenario...
SIP call comes in and gets answered by extension A (MixMonitor is
executed as part of this inbound dial plan of the number being
On Tue, 2011-08-02 at 10:58 +0100, Ishfaq Malik wrote:
Hi
I'm using asterisk 1.8.3.2 (with a couple of patches)
I have the following scenario...
SIP call comes in and gets answered by extension A (MixMonitor is
executed as part of this inbound dial plan of the number being called)
I'm trying to setup SMS among users of a single asterisk box.
I've set up asterisk10-beta to send SMS messages using MessageSend(). If I
manually set the 'from' variable. I can two way messages only between those
two extensions.
i.e.
[sms]
exten = 12000,1,MessageSend(sip:12000,12001)
exten =
I am happy it's being taken care of.
Would the patch fix systems that used the Repo to install Asterisk 1.6.2.19?
That is where we all have problems. Or maybe a new version of Asterisk which
yum update would do the job?
On Mon, Aug 1, 2011 at 11:27 PM, Jonathan Rose jr...@digium.com wrote:
We
I would like to try the ILBC codec on one of our systems.
The system is currently running Asterisk 1.8.5.0 installed from the
Asterisk provided repositories for Centos 5.
Is there a process for installing the ILBC codec under this
environment, or will I have to un-install the RPMs and build
On 08/02/2011 11:42 AM, Bob Pierce wrote:
I would like to try the ILBC codec on one of our systems.
The system is currently running Asterisk 1.8.5.0 installed from the
Asterisk provided repositories for Centos 5.
Is there a process for installing the ILBC codec under this
environment, or will
Hi,
I´ve compiled asterisk-1.8.5.0 on my Debian based distro (Pinguy)
I also compiled iksemel (v1.4) with the option 2./configure
--with-libgnutls-prefix=/usr
As explained in this link (to avoid compilation error )
http://code.google.com/p/iksemel/issues/detail?id=29#c3
I configured
I would be very interested in iLBC. I even posted regarding this to this
mailing list and the thread died after no one was able to confirm it works.
I think there are others who would really like to see H.323 working from the
repo as well (I think that is not working as well).
Regards,
On Tue,
If we are talking about adding stuff to the repo I would vote for jabber and
gtalk also fax (spandsp)
-Original Message-
From: Bruce B bruceb...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 2 Aug 2011 13:36:31
To: Asterisk Users Mailing List - Non-Commercial
Running build 1.8.5.0 (compiled from source) I seem to be having an issue
with codec negotiation. I have a Grandstream HT503 FXO port connected to a
pstn line, a Polycom SP501, and a SIP trunk with callwithus.
What I'm essentially looking to accomplish is for ulaw or g729 (preferably
ulaw) to be
Install OpenSSL-devel (or whatever the equivalent ubuntu package is called) and
then recompile / reinstall and test it again.
Thanks,
--Warren Selby, dCAP
On Aug 2, 2011, at 12:06 PM, neo haux neo.h...@gmx.com wrote:
Hi,
I´ve compiled asterisk-1.8.5.0 on my Debian based distro (Pinguy)
On Tue, 2 Aug 2011 11:42:19 -0500
Bob Pierce westman...@gmail.com wrote:
Is there a process for installing the ILBC codec under this
environment, or will I have to un-install the RPMs and build Asterisk
from source?
You can write a short makefile for just codec_ilbc module, build it and
Hi Jonathan,
Any clue with 1.6.2.19.*1 *might be released?
Regards,
On Mon, Aug 1, 2011 at 11:27 PM, Jonathan Rose jr...@digium.com wrote:
We worked on this bug today and are expecting to release packages with the
fix soon, possibly tomorrow (Aug 2). The issue arose from a change in
Can you please point me to the patch that you just made?
Thanks
On Mon, Aug 1, 2011 at 11:27 PM, Jonathan Rose jr...@digium.com wrote:
We worked on this bug today and are expecting to release packages with the
fix soon, possibly tomorrow (Aug 2). The issue arose from a change in
features
On Tue, Aug 2, 2011 at 4:39 PM, Lefteris Zafiris zaf@gmail.com wrote:
You can write a short makefile for just codec_ilbc module, build it and
install it on your running asterisk system. You will have to install the
asterisk18-devel package and get the asterisk source code either from
a tar
There is much more to installing and configuring OOH323 as it's not easy
breezy install. I think a professional developer help would be
more appropriate than users patching. Just my thought.plus it adds a
great deal of functionality to Asterisk to allow for all add-ons to be
install via
Can you please point me to the patch that you just made?
The patch is committed to v1.6.2 SVN branch.
Patch for v1.6.2 only.
r330490 | jrose | 2011-08-01 16:08:10 -0500 (Mon, 01 Aug 2011) | 12 lines
Asterisk 18103 - Fix reload crash caused by destroying default parking lot
Default parking
TE410P card down.
I have three (3) TE410P in one machine running asterisk with SS7.
My problems started last week when one of my cards started switching to E1
every time after reboot. I set the following in dahdi.conf and that solve
the problem.
/etc/modprobe.d/
options wct4xxp
Hi All,
Along with my asterisks server, all incoming calls to
my D-link DPH-80 ip phones are are working fine while calling from soft
phones with good voice clarity. But not able to make outgoing calls from the
same D-link DPH-80 ip phones to either soft phone or IP phone. What would be
Hi,
I am running asterisk 1.8.5.0 and have compiled in the srtp module
All but Snom phones are working.
I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they
worked for a few hours. This morning all snoms are reporting this when trying
to make a call (this is snom calling
On Tue, Aug 02, 2011 at 09:22:46PM -0400, Dave George wrote:
TE410P card down.
I have three (3) TE410P in one machine running asterisk with SS7.
My problems started last week when one of my cards started switching to E1
every time after reboot. I set the following in dahdi.conf and that
Hi,
We opened the server an checked that the cards were seated correctly and
they are. I will have the tech completely remove them tomorrow and try
again. I will post the results.
Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
dmesg:
wct4xxp :0a:03.0: SPAN 9: Primary Sync Source
wct4xxp :0a:03.0: Span 2 configured for ESF/B8ZS
wct4xxp :0a:03.0: SPAN 10: Primary Sync Source
wct4xxp :0a:03.0: All spans in alarm : No validspan to source RCLK from
wct4xxp :0a:03.0: RCLK source set to span 1
wct4xxp
If it doesn't go green when you put a hard loopback on the port, then contact
Digium support.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave George
Sent: Tuesday, August 02, 2011 10:52 PM
To:
On Tue, Aug 02, 2011 at 03:40:17PM -0500, Warren Selby wrote:
Install OpenSSL-devel (or whatever the equivalent ubuntu package is called)
libssl-dev
and then recompile / reinstall and test it again.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
Hi,
I am using Asterisk1.4, in need to configure barge of all SIP Id's.
SIP Id's start from 900 to 999,
configuration for barge using ChanSpy application. in extensions.conf
exten = 81,1,ChanSpy(SIP)
exten = 81,2,Hangup
* for next barge
But problem is at whenever 938 is comming at press * its
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