Would this be of any help to you?
http://lists.digium.com/pipermail/asterisk-users/2011-June/263339.html
[SATISH]
Mumbai, India.
On Mon, Jun 27, 2011 at 7:14 AM, Rafael dos Santos Saraiva
rafaels...@gmail.com wrote:
I am referring to 3-way conference
Att,
Rafael Saraiva
2011/6/26
When i reload asterisk, calendar show calendars does not show this.
What I am missing? I really need to get this to work!
You are missing that you should take out passwords from config files.
Hope your gmail account didn't get hacked.
--
Hi ,
can you any buddy provide agi script in perl or php for,
only working hours incomming calls forward to his cellno., and after working
hours should be play one playback msg then forward voicemail to his
extension.
working hours(sun - thu, 9:00 to 19:00)
ex: dialplan
exten =
On 06/27/2011 05:30 AM, mahesh katta wrote:
can you any buddy provide agi script in perl or php for,
only working hours incomming calls forward to his cellno., and after
working hours should be play one playback msg then forward voicemail to
his extension.
working hours(sun - thu, 9:00 to
Wasn't that helpful?
http://lists.digium.com/pipermail/asterisk-users/2011-June/264082.html
Use GotoIfTime in agi of your choice with condition part being calculated
dynamically as per your requirement.
But I really don't see any usefulness of AGI if your working hours are fixed
i.e. Mon - Thu,
Hello,
While using FFA, how can we program a specific extensions.conf script to
accept the fax in G.711, while for other calls (from a different source), to
allow T.38 with or without fallback?
The current ReceiveFax(filename,f) is not working for us with one of our
providers.
Thanks,
Michael
hi list,
I have a problem with Asterisk 1.8
I installed the software via the yum repositories of asterisk.org but if
I go to the /etc/asterisk/ I do not find any files in it?
possible?
thanks in advance
p
--
_
-- Bandwidth and
Have you installed sample configuration files package?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paolo De Michele
Sent: Monday, June 27, 2011 4:26 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] not find files
the file already installed:
Updating:
asterisk-sounds-core-en-gsm
asterisk18
asterisk18-addons
asterisk18-addons-bluetooth
asterisk18-addons-core
asterisk18-addons-mysql
asterisk18-addons-ooh323
Hello List,
We are facing a problem in broadcasting DTMF from MeetMe.
Our client is sending DTMF with duration=160 (in SIP INFO) to asterisk but
asterisk is changing this header to different values like 162, 175 etc while
broadcasting to all the participants. Is it possible to restrict asterisk
On 06/27/2011 06:11 AM, Michael wrote:
Hello,
While using FFA, how can we program a specific extensions.conf script to
accept the fax in G.711, while for other calls (from a different
source), to allow T.38 with or without fallback?
The current ReceiveFax(filename,f) is not working for us with
Thanks a lot. OK, from where you got these files? I am trying to know the
source so I can get from it any missing file that the phone is needed.
Regards
Bilal
-
bilal ghayyad wrote:
Dears;
The Cisco 7942 worked in SIP and did not work in
skinny firmware (in skinny,
Hi all
How can I change the festival application voice in asterisk from mailvoice
to femailvoice.
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
bilal ghayyad wrote:
from where you got these files?
I found the link on http://www.voip-info.org
I searched for Cisco 7940
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
If you're installing from source you need to do
make samples
On Mon, 2011-06-27 at 13:38 +0200, Paolo De Michele wrote:
the file already installed:
Updating:
asterisk-sounds-core-en-gsm
asterisk18
asterisk18-addons
I solved by installing asterisk18-configs
regards,
p
On 06/27/2011 01:38 PM, Paolo De Michele wrote:
the file already installed:
Updating:
asterisk-sounds-core-en-gsm
asterisk18
asterisk18-addons
Sorry, read your problem properly this time
yum install asterisk18-configs
On Mon, 2011-06-27 at 13:53 +0100, Ishfaq Malik wrote:
If you're installing from source you need to do
make samples
On Mon, 2011-06-27 at 13:38 +0200, Paolo De Michele wrote:
the file already installed:
On Thu, Jun 23, 2011 at 3:12 PM, Tim Panton t...@westhawk.co.uk wrote:
You should probably not mention the voipusersconfere...@gmail.com address
this for week's VUC
as at the moment the gateway ignores any calls to it.
If/when it comes back to life, we can realistically expect wideband
Hi Kevin,
Controlling it through the sip.conf peers is sufficient for us for this case
(because this particular provider doesn't support T.38 at all), but I think
it would be a good idea to add the option to enable/disable T.38 from the
dialplan. If I recall correctly, that's how callweaver
On Mon, Jun 27, 2011 at 9:06 AM, Michael voip.quest...@gmail.com wrote:
Hi Kevin,
Controlling it through the sip.conf peers is sufficient for us for this case
(because this particular provider doesn't support T.38 at all), but I think
it would be a good idea to add the option to
That's not the password.
I switched it to that in the config file for realism.
I always give some honey out to those who have a sugar tooth.
Any ideas on the fix?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
That's not the password.
I switched it to that in the config file for realism.
I always give some honey out to those who have a sugar tooth.
Any ideas on the fix?
On Mon, Jun 27, 2011 at 3:24 AM, Matt Darnell mattdarn...@gmail.com wrote:
When i reload asterisk, calendar show calendars
Hey - Is there an asterisk interface to gstreamer?
I have not found anything...
Perhaps using a web cam and v4l to get some video going on a PC with
asterisk running.
Call come into extensions.conf and take the video and send it to
xvimagesink and audio to alsa
with outgoing video coming
Emailed wrong address
-Original Message-
From: ERIC HERRON [mailto:e...@lanline.com]
Sent: Monday, June 27, 2011 10:58 AM
To: 'asterisk-users@lists.digium.com'
Subject: FW: [asterisk-users] Asterisk 1.8.4 - Google iCal not working
Forgot link..
That's not the password.
I switched
Hi
Since switching from 1.6.x to 1.8.4 I have noticed the following
1. When you do a 'core show channel channel name' the resulting
information only shows data for Frames In , Frames out is always
0.
2. The rtptimeout option in the sip.conf no longer seems to work. I
have this set to 60 seconds
We just finished an upgrade of our Asterisk system to an HA
environment on a pair of servers using Linux-HA. As part of the
upgrade, we also moved to Asterisk version 1.8.4.3
Most things are working quite nicely on the new system. However, I’m
having trouble getting a paging feature to work. In
Hello,
I have question regarding the changes that are made in the sip
protocol in Asterisk - the option progressinband.
When this option is set to yes in asterisk version 1.4.21.1 - the call flow is:
sip.conf:
progressinband=yes
DeviceAsterisk
---INVITE SDP-
On 06/27/2011 08:06 AM, Michael wrote:
Controlling it through the sip.conf peers is sufficient for us for this
case (because this particular provider doesn't support T.38 at all), but
I think it would be a good idea to add the option to enable/disable T.38
from the dialplan. If I recall
You could force g711 inbound by using
Set(SIP_CODEC=ulaw)
-Original Message-
From: Kevin P. Fleming kpflem...@digium.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 27 Jun 2011 14:08:00
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List -
Hi,
I'm fighting with HylaFAX, Asterisk and T38Modem since some time, to get
fax2mail and mail2fax working, my SIP-provider only supports T38 and
thus using G711 with IAXModem is not an option.
I have got running mail2fax with HylaFAX+ 5.5.0, Asterisk 1.4.20 and
T38Modem 2.0.0 successfull, but
Call file are not suitable for you as asterisk process these files in serial
mode (single threaded) and in case of large number of files processing of
last file can be that much delayed that some portion of message may be
already played or the 1st phone may be hanged.
-Original Message-
31 matches
Mail list logo
