For what channel driver, and what use case?
It's my understanding that in the traditional telephone network
(ISDN/SS7/analog), prior to a call being answered, you were not necessarily
guaranteed a two way media path. Sometimes it was available (there are few
stories of large companies who
Hi,
Asterisk unable to receive DTMF tone from sip client.
Im using the (d) flag in dial application to perfume one digit exit during
ringing state. But unfortunately doesn't work.
Here is my sip configuration :-
[100]
type=friend
username=100
host=dynamic
nat=yes
canreinvite=no
allow=all
Hi,
How to accept DMTF tone during ringing mode? Its possible.
Regards
-Hadi.Salem
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> On Mon, Nov 2, 2015 at 3:16 PM, hadi <almarzuki2...@hotmail.com> wrote:
> > I have configure bridgeConference. But im having some issue. I want to
> > give the ability to the user when dialing from the Conference to
> > hangup the call by sending dtmf tone
I have configure bridgeConference. But im having some issue. I want to give
the ability to the user when dialing from the Conference to hangup the call
by sending dtmf tones without being hangup from the Conference. For example
if the user call some person and that person not answering, the user
(to prevent spoofing).
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi
Sent: Friday, July 31, 2015 11:11 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] showing sip number
trying to do caller ID
blocking
to a regular number and it works too, as long as it¹s a typical long
distance or
local call.
On 7/31/15, 12:10 PM, asterisk-users-boun...@lists.digium.com on behalf
of hadi asterisk-users-boun...@lists.digium.com on behalf of
almarzuki2...@hotmail.com wrote
Hi,
I have asterisk installed on centos with phpagi. Also I have PRI card
connect to it. it's possible to show the sip number when calling from sip
number to external number thru the PRI, instead of showing the PRI number
show the sip number ?
Regards
-Hadi.Salem
--
Hi,
I just want to confirm that my problem is solved now and everything is
working as expected .
I used the patch provided in the following link:
https://reviewboard.asterisk.org/r/2171/
Special thanks to Asterisk development team for great responsibility and
quick reaction.
regards
tried mysql real time module but since I am working with some websocket
clients I have some issues to forward calls from udp clients to websocket
ones.
*I am working with trunk asterisk 11 (r 373330 ) and I tried it with the
latest trunk and same results .
Regards
Hadi Ams
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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Dear All
I need to offer dialup connection for my subscribers. When I put the codec
on G.711 the dialup connection will be successful but for the G.723 G.729
it is not. Can you please let me know what are stuffs do I need to have
dialup connection when choosing G.723 G.729 codecs?
Thank you
--
On Wed, Feb 3, 2010 at 12:17 AM, Ben Dinnerville b...@voicelogic.com.auwrote:
This is usually due to an error with the SIP stack not being loaded due
to an error - make sure that full logging is on and check your log file
and search for ERROR and see if there is any mention to SIP (chan_sip.o
Dear All
On my CentOS 5 server , I have upgraded my Asterisk from 1.4 to 1.6.2 but
its CLI help does not show sip and when dialing outward sip it complains as
'sip not implemented' . Can you please let me know what is wrong my case
here ?
Thank you
--
On Tue, Feb 2, 2010 at 12:40 PM, hadi motamedi motamed...@gmail.com wrote:
Dear All
On my CentOS 5 server , I have upgraded my Asterisk from 1.4 to 1.6.2 but
its CLI help does not show sip and when dialing outward sip it complains as
'sip not implemented' . Can you please let me know what
On Wed, Jan 13, 2010 at 7:36 AM, Olle E. Johansson o...@edvina.net wrote:
13 jan 2010 kl. 06.56 skrev hadi motamedi:
Dear All
I have Asterisk 1.4 installed on my Debian server . I am considering
upgrading my Asterisk to the latest version (1.6) . Can you please let me
know what
On Wed, Jan 13, 2010 at 8:27 AM, Olle E. Johansson o...@edvina.net wrote:
My apologies for the multiple copies.
Had issues with a mailserver that somehow wasn't talking to DNS properly.
Now fixed. It behaved like Asterisk does sometimes, very poor when it can't
connect to DNS. Had power
On Wed, Jan 13, 2010 at 8:49 AM, Olle E. Johansson o...@edvina.net wrote:
13 jan 2010 kl. 09.26 skrev hadi motamedi:
On Wed, Jan 13, 2010 at 7:36 AM, Olle E. Johansson o...@edvina.net
wrote:
13 jan 2010 kl. 06.56 skrev hadi motamedi:
Dear All
I have Asterisk 1.4 installed
Dear All
I have Asterisk 1.4 installed on my Debian server . I am considering
upgrading my Asterisk to the latest version (1.6) . Can you please let me
know what are the major benefits when upgrading from Asterisk 1.4 to
Asterisk 1.6 ?
Thank you
--
Dear All
You are not willing to help me anymore ?
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On Sun, Jan 10, 2010 at 10:58 AM, Gergo Csibra csi...@gmail.com wrote:
Sunday, January 10, 2010, 11:24:22 AM, hadi wrote:
You are not willing to help me anymore ?
Why do you think this?
--
Best regards,
Gergomailto:csi...@gmail.com
On Sun, Jan 10, 2010 at 2:43 PM, Geoff Lane ge...@gjctech.co.uk wrote:
On Sunday, January 10, 2010, Francesco Peeters wrote:
Yes, post your question clear and consicely, include all relevant
information and snip all unneccessary history.
Note that: no reply != not wanting to help...
It
On Mon, Jan 11, 2010 at 6:23 AM, Zhang Shukun bit...@gmail.com wrote:
you'd better paste your dialplan snip here, in order to get specific help.
2010/1/11 Darrick Hartman dhart...@djhsolutions.com:
On 01/10/2010 11:38 PM, hadi motamedi wrote:
FWIW, he did post his question yesterday
Dear All
My Asterisk has sip connection with an external sip server
@192.168.0.139. I have sip inbound and outbound calls as ok . But
there is a problem on
sip incoming calls . To illustrate the problem , please suppose the sip
phone on external sip server dials my Asterisk sip phone @6672019 .
Dear All
Can you please let me know how can I define incoming route to accept
incoming calls from an external sip server?
I have defined the following profile for my sip phone :
Under sip.conf :
-
[osaka]
type=friend
context=sip-outgoing
host=192.168.0.139
disallow=all
to the question itself,
On Wed, Jan 06, 2010 at 10:44:31AM +, hadi motamedi wrote:
Can you please let me know how can I define incoming route to accept
incoming calls from an external sip server?
Just send them there?
I have defined the following profile for my sip phone :
Under sip.conf
--
Best Regards
Yawar Hadi Noshahi
Consultant/Software Engineer
NGI Islamabad
MS Computer Science
Linkoping University
Sweden
+46700-445479
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asterisk-users mailing
On Tue, Jan 5, 2010 at 8:59 AM, Olle E. Johansson o...@edvina.net wrote:
4 jan 2010 kl. 14.46 skrev Kevin P. Fleming:
hadi motamedi wrote:
Sorry . I didn't get the point clearly . In the SIP Invite message , it
says my audio endpoint is IP x.x.x.x port x, and I can use codecs
A,B,C
On Mon, Jan 4, 2010 at 1:46 PM, Kevin P. Fleming kpflem...@digium.comwrote:
hadi motamedi wrote:
Sorry . I didn't get the point clearly . In the SIP Invite message , it
says my audio endpoint is IP x.x.x.x port x, and I can use codecs
A,B,C. The remote endpoint responds with a 200 OK
Dear All
Further to my previous inquiry regarding Asterisk sending dialed digits in
one-by-one digit format when we had ISDN PRI link with the PSTN switch , you
told me that we are expected to enable overlap dialing . At now , we have
the same configuration but sip connection to the external sip
Dear All
Can you please give me guidelines and the link to join Asterisk real time
chat to have your online technical support?
Thank you
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On Thu, Dec 31, 2009 at 12:10 PM, Kevin P. Fleming kpflem...@digium.comwrote:
hadi motamedi wrote:
Can you please let me know if we can have different codec schemes for
audio codec in audio codec out ? I mean , in one application , we
can have our audio codec input set to G.711 a-law
On Thu, Dec 31, 2009 at 6:40 AM, hadi motamedi motamed...@gmail.com wrote:
Dear All
Please be informed that my Asterisk has sip connection to an external
sip server but the sip outgoing call will be disconnected for some
unknown reasons . Please find attached the debug log . Can you please
Dear All
I want to enable festival text-to-speech . To this end , I added the
required lines to festival.scm but when I want to start festival server I
face with the following error :
#festival --server
SIOD ERROR: end of file inside list
Closing a file left open: /usr/share/festival/festival.scm
Dear All
Can you please give me more hint on how Asterisk Dictate() works?
Thank you
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Dear All
Can you please let me know if we can have different codec schemes for
audio codec in audio codec out ? I mean , in one application , we
can have our audio codec input set to G.711 a-law and our audio codec
output set to G.711 u-law . I am facing with an application that calls
for such a
On Wed, Sep 9, 2009 at 4:02 AM, Jeff LaCoursiere j...@jeff.net wrote:
On Wed, 9 Sep 2009, hadi motamedi wrote:
Thank you for your message . But I tried to find it on my server , as the
followings :
#find / -name sip.cfg -print
But it didn't return any result . Can you please let me
] *On Behalf Of *hadi motamedi
*Sent:* 22 December 2009 10:47
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Inquiry:Installing Asterisk 1.4 on CentOS
5.2?
On Tue, Dec 22, 2009 at 9:23 AM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
On Tue
On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner f...@teamforrest.com wrote:
On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote:
On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com
wrote:
On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote:
Dear All
I have
On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham
dcunning...@voisonics.com wrote:
Hadi,
You could use Asterisk as a sip server, it's installable on Windows.
Using sip set debug on might help you with the Host '192.168.0.139' does
not implement 'REGISTER' problem.
On Wed, Dec 23, 2009
On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham
dcunning...@voisonics.com wrote:
Hadi,
You could use Asterisk as a sip server, it's installable on Windows.
Using sip set debug on might help you with the Host '192.168.0.139' does
not implement 'REGISTER' problem.
On Wed, Dec 23, 2009
.
Although I am wondering how much help all this will be in debugging a
connection problem to another SIP provider...
On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi motamed...@gmail.comwrote:
On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham
dcunning...@voisonics.com wrote:
Hadi,
You
On Tue, Dec 22, 2009 at 9:23 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Tue, Dec 22, 2009 at 07:37:50AM +, hadi motamedi wrote:
On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby wcse...@selbytech.com
wrote:
And what is the output of the ./configure? Does it generate any
Dear All
I have tried to install the asterisk-1.4 , libpri-1.4 , and zaptel-1.4 on my
CentOS 5.2 server , but my installation unsuccessful . When I check for the
presence of installed packages , like the followings , I see the output for
libpri and zaptel but nothing is seen for asterisk :
On Mon, Dec 21, 2009 at 12:51 PM, Dan Journo
d...@keshercommunications.comwrote:
Do you have any error logs? What output do you get when you try “make
install” with the asterisk package?
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On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby wcse...@selbytech.com wrote:
On Mon, Dec 21, 2009 at 11:12 PM, hadi motamedi motamed...@gmail.comwrote:
Please find below the error message that I got when issuing make install
:
[r...@mss-0 asterisk-1.4.26]# make install
make: -F.: Command
On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby wcse...@selbytech.com wrote:
And what is the output of the ./configure? Does it generate any errors?
Thanks,
--Warren Selby
On Dec 22, 2009, at 1:09 AM, hadi motamedi motamed...@gmail.com wrote:
On Tue, Dec 22, 2009 at 6:56 AM, Warren
Dear All
I have an application that calls for my Asterisk sip to be connected to an
external sip server for voip routing . Please be informed that my Asterisk
sip is at @192.168.0.2 and the external sip is at @192.168.0.139 . To this
end , I modified my sip.conf extensions.conf as the followings
On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner f...@teamforrest.com wrote:
On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote:
Dear All
I have an application that calls for my Asterisk sip to be connected to
an external sip server for voip routing . Please be informed that my
Asterisk sip
Dear All
I have an application that calls for Asterisk sip configuration to be able
to communicate with external sip server . My Asterisk 3.1.14 has been
installed on Debian 3.1 server and the external sip server is
@192.168.0.10, the same subnet as my Debian server @
192.168.0.2 . At now , the
Yawar Hadi Noshahi
Consultant/Software Engineer
NGI Islamabad
MS Computer Science
Linkoping University
Sweden
+46700-445479
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main contrib
Thank you in advance
On Sun, Nov 15, 2009 at 6:36 AM, Jarrod Lash jar...@fed-com.com wrote:
you are running a old version of debian?
what repository are you using (cat /etc/apt/sources.list)?
On Sun, Nov 15, 2009 at 1:27 AM, hadi motamedi motamed...@gmail.comwrote:
Sorry . I
Dear All
Can you please do me favor and let me have the link to download the Asterisk
1.4.13 for my Debian server ? Please let me know how to install it .
Thank you in advance
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Dear All
Please be informed that I need to install Asterisk 1.4.13 on my Debian 3.1
server . But I got the following message when trying for #./configure :
error: no acceptable C compiler found in $PATH
Can you please do me favor and let me know what is the problem ?
Let me thank you in advance
update
then
apt-get install gcc g++
--
Jarrod Lash, jar...@fed-com.com
Federated Communications
www.fed-com.com
Office: +1-412-357-2127
Mobile: +1-412-999-0049
Fax: +1-412-545-8368
On Sun, Nov 15, 2009 at 12:31 AM, hadi motamedi motamed...@gmail.comwrote:
Dear All
Please
Dear All
Can you please do me favor and let me know how can I stop my Asterisk ? Can
you please confirm if the following procedure is correct to stop it ?
#/etc/init.d/asterisk stop
#cd /etc/init.d
#chmod asterisk
Let me thank you in advance
___
--
cli stop now
or
cli stop gracefully
:)
otherwise
pkill -9 asterisk
On Sat, Nov 14, 2009 at 7:39 AM, hadi motamedi motamed...@gmail.com wrote:
Dear All
Can you please do me favor and let me know how can I stop my Asterisk ? Can
you please confirm if the following procedure is correct to stop
Dear All
Can you please do me favor and let me know if there is an facility in
Asterisk server that can be used to have remote access to the server ?
Please be informed that we have installed commissioned our Asterisk server
at remote site with DECT telephony service provisioning for our
Thank you for your reply . But I am seeking for PPPoE remote access that
fits my case here . Can you please let me know if there is any solution in
this regard ? (like PPPD)
On Sat, Sep 26, 2009 at 12:16 PM, Michiel van Baak mich...@vanbaak.infowrote:
On 09:41, Sat 26 Sep 09, hadi motamedi
...@gmail.com wrote:
use
Asterisk now software. You can access by IP.
On Sat, Sep 26, 2009 at 2:11 PM, hadi motamedi motamed...@gmail.comwrote:
Dear All
Can you please do me favor and let me know if there is an facility in
Asterisk server that can be used to have remote access
:
A good way is to give try
On Sat, Sep 26, 2009 at 11:41 AM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
yeah it can :)
On Sat, Sep 26, 2009 at 11:30 AM, hadi motamedi motamed...@gmail.comwrote:
Thank you for your reply . Excuse me , you mean the Asterisk can play SLN
files ? Can you
Dear All
Can you please do me favor and let me know which Asterisk codec you will
prefer when you want to offer your subscribers with dialup data connection ?
Let me thank you in advance
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at 1:29 PM, hadi motamedi motamed...@gmail.comwrote:
Dear All
Can you please do me favor and let me know why my converted sound files
are not being played and heared on my Asterisk ? Please find attached my
sound files . Actually , I had them recorded as *.wav files and I tried to
convert them
...@gmail.com wrote:
check the file formats first if .wav is listed there and if it is, then
check the translation if its activated.
On Tue, Sep 8, 2009 at 2:05 PM, hadi motamedi motamed...@gmail.comwrote:
No . I don't receive any error message after converting from *.wav to
*.gsm but the new
Dear All
I sent you a message regarding my problem with Asterisk Call Parking feature
and you told me that needs to check the polycom sip.cfg file . But my
Asterisk doesn't have sip.cfg file . Can you please let me know how can I
overcome ?
___
--
there, then check the translation if you have the codec activated, it
worked for me before.
On Tue, Sep 8, 2009 at 2:40 PM, hadi motamedi motamed...@gmail.comwrote:
Thank you . Please be informed that the *.wav files cannot be played on my
Asterisk so I had to convert to *.gsm file format .I
wrote:
is there an error on the asterisk cli when you're playing the sound file?
On Tue, Sep 8, 2009 at 4:19 PM, hadi motamedi motamed...@gmail.comwrote:
Thank you . Please find below my original and converted sound files
attributes on my Asterisk :
#file FR1.wav
FR1.wav: RIFF
2009, hadi motamedi wrote:
I sent you a message regarding my problem with Asterisk Call Parking
feature
and you told me that needs to check the polycom sip.cfg file . But my
Asterisk doesn't have sip.cfg file . Can you please let me know how can I
overcome ?
sip.cfg is not an Asterisk
Sorry , I checked on my Asterisk pbx and there is no sip.cfg file on it .
Can you please let me know how can I make my Asterisk Call Parking as
functional ?
On Tue, Sep 1, 2009 at 6:23 AM, Lee, John (Sydney)
john@compuware.comwrote:
Please find attached my Asterisk sip.conf .
Can you
Dear All
Can you please do me favor and let me know why my converted sound files are
not being played and heared on my Asterisk ? Please find attached my sound
files . Actually , I had them recorded as *.wav files and I tried to convert
them to *.gsm as the followings :
#sox FR3.wav
should change all of the passwords that are in that file
and yes, change the passwords in all your phones.
Lyle Giese
LCR Computer Services, Inc.
hadi motamedi wrote:
Thank you for your reply . Please find attached my Asterisk sip.conf .
Can you please let me know what modifications
Dear All
Can you please do me favor and let me know what is my problem with my
Asterisk VoiceMail configuration as it doesn't work correctly in my case ?
Please find below that part of my extensions.conf that I intend to make use
of voice mail for No Answer reply :
[line-incoming]
exten =
...@venturevoip.com wrote:
On 1/09/09 6:14 PM, hadi motamedi wrote:
exten = s,n,noop(${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1)
As you see , I intend to redirect the calling party to the called party
voice mailbox if he doesn't answer the call (that will be set at the
number
...@venturevoip.com wrote:
On 31/08/09 5:49 PM, 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 phone type is ANABELL phone .
Please do me
:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *hadi motamedi
*Sent:* Monday, August 31, 2009 1:09 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Inquiry:How to hide Caller Id
Thank you for your reply . Yes , he is seeing his own
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 phone type is ANABELL phone .
Please do me favor and let me know how can I disable
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
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 2:39 PM
To: Asterisk Users Mailing List - Non-Commercial
Dear All
Can you please do me favor and let me know how I can hide the subs number
being displayed on his phone when he goes off hook ? I mean when the subs
goes off hook he sees his assigned number on his phone and I need to disable
this feature . I don't know from which configuration file this
?
Looking forward your reply
Regards
H.Motamedi
On Mon, Aug 31, 2009 at 6:28 AM, Matt Riddell li...@venturevoip.com wrote:
On 31/08/09 5:24 PM, hadi motamedi wrote:
Dear All
Can you please do me favor and let me know how I can hide the subs
number being displayed on his phone when he goes off
Sorry . I meant subscriber .
On Mon, Aug 31, 2009 at 6:31 AM, Paul Hales pdha...@optusnet.com.au wrote:
Matt Riddell wrote:
What is a subs?
A submarine. I think.
PaulH
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...@gmail.comwrote:
On Thu, Jul 30, 2009 at 1:19 AM, hadi motamedimotamed...@gmail.com
wrote:
Thank you very much for your reply . But please be informed that our
current
line-outgoing route is being configured as the followings (in
extensions.conf):
Set(TIMEOUT(digit)=timeout)
There's
Dear All
Please be informed that we have an application for our subs to be able to
dial #21 to reach IN services . Can you please let us know how we can
support for this as it seems that the Asterisk does not support for the hash
# key as an valid extension to be dialed by the user ?
Regards
Dear All
Can you please let us know how to configure Asterisk to recognize extensions
starting with the hash key ?
Regards
H.Motamedi
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AstriCon 2009 - October 13 - 15 Phoenix,
...@gmail.comwrote:
On Tue, Jul 28, 2009 at 1:01 AM, hadi motamedimotamed...@gmail.com
wrote:
Dear All
Can you please let us know how we can modify our Asterisk inter digit
delay
? Actually , our subs dials his intended numbers with some delay in
between
entering the digits sequentially
Thank you very much for your reply . But please be informed that our current
line-outgoing route is being configured as the followings (in
extensions.conf):
[line-outgoing]
exten = _X.,1,macro(dialuser,Zap/g1/${EXTEN},${EXTEN})
As you see , it is trying to consider the dialed number as an
Dear All
Can you please let us know how we can modify our outgoing extension routing
such that our subs can dial as *21 for reaching to IN services . Please
find below our current item for outgoing dialing , as the followings :
[line-outgoing]
exten =
Dear All
Regarding our opened case , can you please confirm if our attached
extensions.conf file can fullfil the needs of sending the subs dialed digits
one-by-one instead of sending it as an whole packet ?
Regards
H.Motamedi
extensions.conf
Description: Binary data
Dear All
Can you please let us know how we can modify our Asterisk inter digit delay
? Actually , our subs dials his intended numbers with some delay in between
entering the digits sequentially . It seems that our Asterisk pbx will wait
for about 2 seconds and if no extra digits are to be entered
Dear All
It seems that our Asterisk pbx announcement files are being stored inside
the /var/lib/asterisk/sounds folder . Can you please let us know what is
the appropriate program to open and hear them on an MS Windows client ?
(e.g. pbx-invalid.gsm)
Regards
H.Motamedi
(not just an
PBX) .
Regards
H.Motamedi
On Wed, Jul 22, 2009 at 12:53 PM, John Novack jnov...@stromberg-carlson.org
wrote:
Curious - Why?
What is the peer switch and why does it have this requirement?
John Novack
hadi motamedi wrote:
Dear All
Can you please let us know how we can modify
Dear Leif
Can you please provide us with more details on this Overlap Dialing
phillosophy ?
Regards
H.Motamedi
On Wed, Jul 22, 2009 at 1:15 PM, Leif Madsen
leif.mad...@asteriskdocs.orgwrote:
John Novack wrote:
Can you please let us know how we can modify our Asterisk
extensions.conf
Dear All
Can you please let us know how we can modify our Asterisk extensions.conf
file so it interprets the subscriber dialed digits in one-by-one digit
manner . At its current configuration , it interprets them in an whole
packet . I mean , say the subscriber dials as 665 so we need
clue?
Regarda
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Best regards
Yawar Hadi
');
hope u get it
On Mon, Feb 23, 2009 at 1:14 PM, michel freiha mich...@gmail.com wrote:
Dear Sir,
Kindly note that the problem is on command $AGI-get_variable('
variablename');
The AGI seems that it's not reading nothing from asterisk
Regards
On Mon, Feb 23, 2009 at 9:26 AM, Yawar Hadi
of you and specially Mr. Yawar hadi for his great assist and
professionalism
Thanks
On Mon, Feb 23, 2009 at 6:48 PM, Steve Edwards
asterisk@sedwards.comwrote:
On Mon, 23 Feb 2009, Yawar Hadi wrote:
so if u want to read extension then supplu variable name like
$myno=$AGI
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Best regards
Yawar Hadi Noshahi
QAU Islamabad
(+92-0300-5504798)
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How ?
On Mon, Feb 23, 2009 at 12:09 PM, Steve Edwards
asterisk@sedwards.comwrote:
On Mon, 23 Feb 2009, Yawar Hadi wrote:
dear steve
any issue u havent replied.?
You have me confused with michel freiha.
Thanks in advance
ohh got it...sorry for miss interpretation
On Mon, Feb 23, 2009 at 12:23 PM, Yawar Hadi yawarh...@gmail.com wrote:
How ?
On Mon, Feb 23, 2009 at 12:09 PM, Steve Edwards asterisk@sedwards.com
wrote:
On Mon, 23 Feb 2009, Yawar Hadi wrote:
dear steve
any issue u
wrote:
Dear Yawar,
I need please some help from you regarding the script tha you already
provided to me...It seems that the perl script is not reading correctly
variables from asterisk server..Can you please help in that?
Regards
On Mon, Feb 23, 2009 at 8:28 AM, Yawar Hadi yawarh...@gmail.com
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