That doesn't happen on all phones. Either find a way to block that
feature on the phone, or change phones for that location.
I assume you don't want the user to know that phone's local number.
Cary Fitch
_
From: asterisk-users-boun...@lists.digium.com
I couldn't find any information on this brand of phone on the internet
at all.
PaulH
hadi motamedi wrote:
Sorry for lack of enough information . I mean my subscriber when goes
off hook he will see his own number displayed on his phone . I need to
disable this feature on my Asterisk .The
On Monday 31 August 2009 12:56:32 pm Steve Totaro wrote:
On Sun, Aug 30, 2009 at 5:57 AM, Michael Maxwell
metalmic...@gmail.comwrote:
When a signal is *not* being received and or transmitted by the radio
system
attached to Asterisk/app_rpt via its interface, is the incoming and or
I am following this procedure
ou have to compile asterisk with the cdr_pgsql.o module, for this follow the
steps:
Configure asterisk with postgresql support:
./configure --with-postgres=dir where postgresql is installed
Then issue the command:
make menuconfig
in the menu select 2.Call Detail
Thank you for your reply . Yes , he is seeing his own number on his phone
upon going off hook and before dialing any number . Can you please do me
favor and confirm if it is not a feature of Asterisk that I can disable it ?
Regards
H.Motamedi
On Mon, Aug 31, 2009 at 6:53 AM, Matt Riddell
I just tried it on 3 different numbers. Dialed as 10 digits NPANXX
I was told I am sorry but you can only dial within North America..etc.
C Fitch
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan
hello,
i am using following SIPP command to test My meetme conference
./sipp -sn uac -d 30 -s 8600 127.0.0.1 -l 20
which generates 20 call to my server but i need to give delay between each
call
once 1 st call is placed then second call should be placed after few seconds
and is there
Thank you for your reply . I really don't want the user to know the phone's
local number . Can you please do me favor and propose one of the available
phones that doesn't have this feature ?
Regards
H.Motamedi
On Mon, Aug 31, 2009 at 7:12 AM, Cary Fitch ca...@usawide.net wrote:
That doesn’t
Sorry for mis-typing in phone type . Please be informed that the current
phone type our subscribers are using is TP6000 ones .
On Mon, Aug 31, 2009 at 7:24 AM, Paul Hales pdha...@optusnet.com.au wrote:
I couldn't find any information on this brand of phone on the internet
at all.
PaulH
-r is a flag that regulates the call setup rate per second.
DHAVAL INDRODIYA wrote:
hello,
i am using following SIPP command to test My meetme conference
./sipp -sn uac -d 30 -s 8600 127.0.0.1 -l 20
which generates 20 call to my server but i need to give delay between
each call
A Google of that model showed a discontinued Telstra corded phone.
But in any case SNOM and Grandstream phones Do show the number before you
pick up the handset.
I would suggest you use a Grandstream 286 voip adapter and a standard corded
or wireless phone so that the caller doesn't have
Hi,
SNOM dosent show the number, it shows user realname.
http://wiki.snom.com/wiki/index.php/Settings/user_realname
// Magnus
Från: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] För Cary Fitch
Skickat: den 31 augusti 2009 09:06
Till: 'Asterisk Users
Lefteris Zafiris schrieb:
I have written a simple application for asterisk 1.6 that uses the Flite
tts engine to render text to speech.
Source is available here: http://zaf.github.com/Asterisk-Flite/
It works more or less like the festival app, can use cache etc.
Its only tested against
thanks Alex,
it works
but can you tell me about any sound playing on SIPP means ,
once SIPP channels connect in conference room then there is lots of noise ,
is there any way to reduce it.
regards
Dhaval
On Mon, Aug 31, 2009 at 12:38 PM, Alex Balashov
abalas...@evaristesys.comwrote:
-r is
Olle E. Johansson schrieb:
27 aug 2009 kl. 11.24 skrev Klaus Darilion:
Hi!
I want to use Asterisk as load generator to test quality degradation
with increased load (e.g. testing other SIP equipment or IP-links).
Is anybody aware of such a setup with Asterisk - is it possible to get
RTP
On 31/08/09 8:47 PM, Klaus Darilion wrote:
Olle E. Johansson schrieb:
27 aug 2009 kl. 11.24 skrev Klaus Darilion:
Hi!
I want to use Asterisk as load generator to test quality degradation
with increased load (e.g. testing other SIP equipment or IP-links).
Is anybody aware of such a setup
Klaus Darilion wrote:
Lefteris Zafiris schrieb:
I have written a simple application for asterisk 1.6 that uses the Flite
tts engine to render text to speech.
Source is available here: http://zaf.github.com/Asterisk-Flite/
It works more or less like the festival app, can use cache etc.
Its
Hi,
Is there anyone there that installed successfully the CRI package and
manages to play the calls listed in the call monitor page?
Regards.
Original Message
Subject: Re: [asterisk-users] Crystal Recording Interface
From: Cyprus VoIP voi...@gmail.com
To: Asterisk Users
Hi
I am using asterisk version 1.6.0.5
I have build up one utility that will fire Originate Action on Manager...
In which, i have define number to call eg. 919912312345 (MobileNumber)
How can i know that this number format is true for Indian Number...
In originate action, user can enter any
Use the pattern matching P137 in Asterisk: the future of telephony
For example
Exten = 919X,n,
Cordialement,
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De
Use the pattern matching P137 in Asterisk: the future of telephony
For example
Exten = _919X,n,
Cordialement,
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Hello,
I am trying to setup an asterisk box for a small office that has 4 phone
lines and a fax. The fax will not be going through the box. I have Digium
TDM410P to take 4 analog lines and I will be using grandstream gxp2000 for
our setup. I have read the docs just do not understand the
Thanks Matt, and everybody else, very useful information. I guess I'll have
to sit again and spend time coding delays, small amount payments for new
accounts and paypal=signup email match.
--
Zeeshan A Zakaria
On Mon, Aug 31, 2009 at 12:07 AM, Matt Riddell li...@venturevoip.comwrote:
On
Cary, thanks for your feedback. You tried dialing directory assistance
numbers which cost dollar a minute. They can't be free. But you got the
voice messages, which means the webphone worked for you.
--
Zeeshan A Zakaria
On Mon, Aug 31, 2009 at 2:18 AM, Cary Fitch ca...@usawide.net wrote:
I
Topic 6: RE:unable to execute command
hi there
i tried to execute the command as suggest like
exten = 1987,1,Playback(posix-restarting)
exten = 1987,2,wait(1)
exten = 1987,3,System(/usr/bin/python /home/docas/Desktop/mess1.py)
exten= 1987,4,Hangup
it still doesn't work,now it does it give
Matt Riddell schrieb:
On 31/08/09 8:47 PM, Klaus Darilion wrote:
Olle E. Johansson schrieb:
27 aug 2009 kl. 11.24 skrev Klaus Darilion:
Hi!
I want to use Asterisk as load generator to test quality degradation
with increased load (e.g. testing other SIP equipment or IP-links).
Is
I've done gain tuning as per the info I've found online. I've got my
RXGain set so my volumes list as about 14,800 (using a milliwatt test
number and ztmonitor -vv). However listening to the line now, this
sounds too loud to me. The person speaking sounds fine, but I've now
got a large
When you start taking credit card payments (assuming you will), be
careful with small payment amounts. You'll become a fraud haven. A lot
of CC thieves or people who've just bought a CC number will use a small
amount charge to check and see if the card is any good.
Check out some of the MaxMind
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
ABBAS SHAKEEL wrote:
but when i execute this ./configure --with-postgres=dir where
postgresql is installed
it gives an error for missing an pg_config file . i searched the PC
but it really dont exists. but database server is fine and
Hi,
We are trying to implement a complex business logic in Asterisk. Executing
Wait_For_Digit command after playing IVR. We want to stop the IVR once we
receive the digit. It is not recognizing the Digit until it completes the
IVR. How can we stop the IVR once we receive the digit?
Thanks
I didn't know about MaxMind, but seems like a great service. Seems like
exactly what I needed. Thanks for the reference.
--
Zeeshan A Zakaria
On Mon, Aug 31, 2009 at 8:59 AM, SIP s...@arcdiv.com wrote:
When you start taking credit card payments (assuming you will), be
careful with small
Are you using Background(SomeSoundFile) ?
or PlayBack(SomeSoundFile) ?
Normally Background() will stop if the pressed digit(s) match any
dialplan entry.
Bharath B. Reddy Bynagari wrote:
Hi,
We are trying to implement a complex business logic in Asterisk.
Executing “Wait_For_Digit”
Use read with one digit. If you want a specific digit, test for it like
this
exten = s,1(readacct),Read(digitacc,record/enteracct,1,skip,1,5])
exten = s,n,Gotoif($[${LEN(${digitacc})} 1]?readacct)
exten = s,n,Gotoif($[${digitacc} != 5]?readacct)
This instance loops until 5 is pressed.
I think the IVR audio must be playing in Background mode, not Play Mode.
Try that. Background means play the sound and move on to the next
instruction. Play means to play the sound and after it is over, move to the
next instruction.
Cary Fitch
_
From:
This is correct. You can build a pretty large voice menu using
Background(file1file2file3file4) stacked up with the knowledge that the
user can skip all of it by pressing a key.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
How to extract that CDR from asterisk ?
On Sat, Aug 29, 2009 at 2:11 AM, David @ULC ucoms2...@gmail.com wrote:
From Asterisk, I need a List of Numbers , asterisk dialed out.
I am looking for status of each number dialed out.
Whether its failed or successful .
Any way ?
Depends on your setup. It's either a table in a database or
/var/log/asterisk/cdr-csv/Master.csv (or both).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC
Sent: Monday, August 31, 2009 9:49 AM
To:
Hi folks!
I have an asterisk, version 1.4.22.2, with dahdi 2.1.0.4 and OpenR2 2.1.1.0.
Sometimes twice a day, sometimes after 3 days, all sip devices looses
registry, but asterisk doesn't show nothing strange, no error log, and
all calls in E1 trunk continue running, but sending to voicemail.
It sounds like your SIP devices aren't set up to periodically(frequently)
re-register themselves. You can resolve this on the device level or have
asterisk poll them for re-registration. It could also be some sort of
firewall/NAT problem that chops the connections at some interval.
On Monday 31 August 2009 08:48:32 am Cary Fitch wrote:
I think the IVR audio must be playing in Background mode, not Play
Mode.
Try that. Background means play the sound and move on to the next
instruction. Play means to play the sound and after it is over, move to the
next instruction.
At 11:56 PM 8/30/2009, you wrote:
Sorry for mis-typing in phone type . Please be informed that the
current phone type our subscribers are using is TP6000 ones .
The phone only knows the number to display if you tell it the number,
so tell it to display something else.
Ira
Hello,
I see that there's 1.6.0.x and 1.6.1.y versions of Asterisk.
Is there a clear table that describes the features and/or differences
between them?
Are both stable enough?
Is T.38 Fax supported on both? If yes, which spandsp is supported? I saw
on voip-info.org that version 6 is not
I set mine at 300 (5 minutes). You might want a higher value if you have
lots of phones, but since I only have 8 at my shop, this causes no
noticeable downside.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
On Mon, Aug 31, 2009 at 12:37 PM, Cyprus VoIPvoi...@gmail.com wrote:
Are both stable enough?
Stable enough for whom? You are the only one who can answer that question.
Is T.38 Fax supported on both?
Yes, with caveats. There continue to be a number of T.38 patches going
into various releases.
Ok Got it.
Any 3rd party Interface which can get me all these result in a front end ?
On Mon, Aug 31, 2009 at 8:19 PM, David @ULC ucoms2...@gmail.com wrote:
How to extract that CDR from asterisk ?
On Sat, Aug 29, 2009 at 2:11 AM, David @ULC ucoms2...@gmail.com wrote:
From Asterisk, I need
Danny,
Thank for your reply.
I'm sure that is not firewall/nat because all sip devices are using a
private class of IP and asterisk has a network adapter with an IP from
the same class/network.
How muchi is a good value for re-register?
Thanks a lot
2009/8/31 Danny Nicholas
On Aug 30, 2009, at 7:45 PM, Zeeshan Zakaria wrote:
I charge my customers through PayPal, but recently faced a fraud
which previously had only heard about. Somebody registered a few
accounts, paid online with paypal (as my service is only prepaid)
and started making expensive long
Hi,
I am using Asterisk personally at home.
My SIP client (SPA 3000) supports MWI with SIP NOTIFY messages.
With a previous version of Asterisk I had no problems with MWI. But now I am
using the following version which comes with Trixbox 2.8.0.1, and I have
problems with MWI.
Asterisk
Good idea Eric regarding welcome package.
--
Zeeshan A Zakaria
On Mon, Aug 31, 2009 at 1:07 PM, Eric Chamberlain e...@rf.com wrote:
On Aug 30, 2009, at 7:45 PM, Zeeshan Zakaria wrote:
I charge my customers through PayPal, but recently faced a fraud
which previously had only heard about.
Cyprus VoIP wrote:
Hello,
I see that there's 1.6.0.x and 1.6.1.y versions of Asterisk.
Is there a clear table that describes the features and/or differences
between them?
Are both stable enough?
Is T.38 Fax supported on both? If yes, which spandsp is supported? I saw
on
Rob a écrit :
Yes ... as a matter of fact here is the sip.conf ... obviously private info
removed
[...]
Did you try to call Gizmo numbers to see if you have success with them?
** Hear your Gizmo5 number repeated back to you.
*0 Test your router's SIP compatibility.
411 The
To view the post and reply , I always to use below link..
http://lists.digium.com/pipermail/asterisk-users/2009-August/thread.htmlhttp://lists.digium.com/pipermail/asterisk-users/2009-February/thread.html
Any better way to access the forum ?
___
--
David @ULC wrote:
Ok Got it.
Any 3rd party Interface which can get me all these result in a front end ?
http://www.areski.net/asterisk-stat-v2/about.php
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither
Hi,
My Trixbox 2.8.0.1 installation includes the following Asterik version:
1.6.0.9-samy-r27
I am having some problems with it and I think they might be solved if I use the
latest Asterisk version.
Is it a good idea to update Asterisk in Trixbox externally ?
Is it safe ?
If so, which version
On Mon, 31 Aug 2009, ilker Aktuna wrote:
Hi,
My Trixbox 2.8.0.1 installation includes the following Asterik version:
1.6.0.9-samy-r27
I am having some problems with it and I think they might be solved if I use
the latest Asterisk version.
Is it a good idea to update Asterisk in Trixbox
Thank you.
That was quick and helpful :)
Then I'll just make and make install
What should I backup, in case of rollback requirement ?
Thanks.
- Original Message -
From: Jeff LaCoursiere j...@jeff.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Mon, 31 Aug 2009, ilker Aktuna wrote:
Thank you.
That was quick and helpful :)
Then I'll just make and make install
What should I backup, in case of rollback requirement ?
That's a bit tougher. At the least /usr/lib/asterisk/modules,
/etc/asterisk, and /usr/sbin/asterisk... someone
One thing I kind of like that Trixbox does is their endpoint manager.
That's about the only feature I haven't been able to replace.
Tom
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
I am a big fan of ubuntu LTS and freepbx and recently I saw mention of a
custom module to add auto configuring endpoints for linksys (but i cna't
find it again right now)
Trixbox had too much stuff whereas the source install of just what you
want is nice and clean
Cheers Duncan
Jeff
I have a _very_ specific situation where I need queues to work in a very
specific manner - I need the queue to only accept one call at a time,
even though several phones are attached to it.
My memory tells me that queues might have even worked this way in the
distant past (pre 1.0)...but I am
Hi,
Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine.
With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a line
polarity reversal. Stutter dialtone is generated as expected.
Has anyone else seen this? Is there anything special I need to do for
1.6.1 to make FSK
Paul Hales escribió:
I have a _very_ specific situation where I need queues to work in a very
specific manner - I need the queue to only accept one call at a time,
even though several phones are attached to it.
My memory tells me that queues might have even worked this way in the
distant
Miguel Molina wrote:
Paul Hales escribió:
I have a _very_ specific situation where I need queues to work in a very
specific manner - I need the queue to only accept one call at a time,
even though several phones are attached to it.
My memory tells me that queues might have even worked
Greetings- I'm wondering if the Digium PRI cards can be used for data (Cisco
HDLC, PPP, etc) or if they're for voice circuits only. I haven't been able to
find any information on this. All documentation direct from Digium seems to
indicate their hardware is for voice applications only.
On Thu, 2009-08-27 at 14:23 -0400, John A. Sullivan III wrote:
Hello, all. In our multi-tenant environment, we would like to be able
to use the reinvite media redirection within Asterisk for calls within a
tenant but not between tenants. We would like inter-tenant calls to be
fully proxied
On Monday 31 August 2009 21:59:28 Tim Nelson wrote:
Greetings- I'm wondering if the Digium PRI cards can be used for data
(Cisco HDLC, PPP, etc) or if they're for voice circuits only. I haven't
been able to find any information on this. All documentation direct from
Digium seems to indicate
Matt Riddell wrote:
On 31/08/09 2:33 PM, Glen wrote:
I have asterisk 1.4.21 and web meetme (latest release 3.1) I have also
installed the latest versions of mysql and php. I followed the readme
file that came with the web meetme app and everything seemed to go fine
up until I realised the
I meant /usr/lib not /var/lib sorry
--
Cheers,
Matt Riddell
Director
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX
On 1/09/09 4:31 PM, Glen wrote:
Matt Riddell wrote:
On 31/08/09 2:33 PM, Glen wrote:
I have asterisk 1.4.21 and web meetme (latest release 3.1) I have also
installed the latest versions of mysql and php. I followed the readme
file that came with the web meetme app and everything seemed to go
Dear All
Can you please do me favor and let me know what is the problem with my
Asterisk call parking as it is not functioning correctly on my Asterisk ?
Please find attached my features.conf . According to my configuration ,
the subscriber needs to press hash (pound) key and dial 700 to initiate
Just a quick guess - is it because you did not program your Polycom digit plan
properly in sip.cfg?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi
Sent: Tuesday, 1 September 2009
Matt Riddell wrote:
On 1/09/09 4:31 PM, Glen wrote:
Matt Riddell wrote:
On 31/08/09 2:33 PM, Glen wrote:
I have asterisk 1.4.21 and web meetme (latest release 3.1) I have also
installed the latest versions of mysql and php. I followed the readme
file that came with the web
On 1/09/09 4:54 PM, Glen wrote:
Parsing '/etc/asterisk/cbmysql.conf': Found
asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_cbmysql.so:
undefined symbol: mysql_init
ldd /usr/lib/asterisk/modules/app_cbmysql.so
This is the output
linux-gate.so.1 = (0xe000)
libpthread.so.0
Thank you for your reply . Please find attached my Asterisk sip.conf . Can
you please let me know what modifications are needed ?
Regards
H.Motamedi
On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney)
john@compuware.comwrote:
Just a quick guess - is it because you did not program your
On 1/09/09 5:08 PM, hadi motamedi wrote:
Thank you for your reply . Please find attached my Asterisk sip.conf .
Can you please let me know what modifications are needed ?
Regards
H.Motamedi
He actually asked for the sip.cfg (i.e. the config for the polycom
rather than for Asterisk)
--
Polycom sip.cfg is not the same as the Asterisk sip.conf file
hadi motamedi wrote:
Thank you for your reply . Please find attached my Asterisk sip.conf .
Can you please let me know what modifications are needed ?
Regards
H.Motamedi
On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney)
On Tue, Sep 1, 2009 at 3:06 PM, Matt Riddell li...@venturevoip.com wrote:
On 1/09/09 4:54 PM, Glen wrote:
Parsing '/etc/asterisk/cbmysql.conf': Found
asterisk: symbol lookup error:
/usr/lib/asterisk/modules/app_cbmysql.so:
undefined symbol: mysql_init
ldd
Please find attached my Asterisk sip.conf .
Can you please let me know what modifications are needed ?
Yes, I am referring to the Polycom sip.cfg and not the sip.cfg in
Asterisk.
Somethere down in sip.cfg, there is a line that looks like this:
digitmap dialplan.digitmap=#700| ...
On 1/09/09 5:19 PM, Glen Ganderton wrote:
app_cbmysql.c:37:1: warning: AST_MODULE redefined
command-line: warning: this is the location of the previous definition
app_cbmysql.c: In function âcheckMaxâ:
app_cbmysql.c:116: warning: implicit declaration of function
âast_say_numberâ
But they do taste similar.
PaulH
Darrick Hartman wrote:
Polycom sip.cfg is not the same as the Asterisk sip.conf file
hadi motamedi wrote:
Thank you for your reply . Please find attached my Asterisk sip.conf .
Can you please let me know what modifications are needed ?
Regards
Hmm, it looks like it has a makefile in the cb_mysql directory which is
supposed to do the linking.
Have you tried running make from there?
It also has a copyright of Mark Spencer, but I know 100% he didn't write it.
The person you're probably looking for is Dan Austin, but I can't track
him
I've sent you Dan Austin's email address off list just in case he is
able to help out :D
--
Cheers,
Matt Riddell
Director
___
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive
In the latest readme for WebMeetMe (3.1.0) it states:
* Compile and install CBMySQL
App_cbmysql is now included in the web-meetme package,
located in ./cbmysql. To install just run make; make install
Copy the sample cbmysql.conf to /etc/asterisk and create
a dialplan similar to
Matt Riddell wrote:
In the latest readme for WebMeetMe (3.1.0) it states:
* Compile and install CBMySQL
App_cbmysql is now included in the web-meetme package,
located in ./cbmysql. To install just run make; make install
Copy the sample cbmysql.conf to /etc/asterisk and create
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