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 Aste
just an
PBX) .
Regards
H.Motamedi
On Wed, Jul 22, 2009 at 12:53 PM, John Novack 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
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
wrote:
>
>
> John Novack wrote:
> >> Can you please let us know how we can modify our Asterisk
> >> "extensions.conf" file so it interprets
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 t
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
_
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 => _X.,1,macro(dialuser,Zap/g1/${EXTEN},${EXT
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
__
Tue, Jul 28, 2009 at 1:01 AM, hadi motamedi
> 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 w
On Thu, Jul 30, 2009 at 1:19 AM, hadi motamedi
> 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)
>
> T
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
H.
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, Arizo
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 fe
?
Looking forward your reply
Regards
H.Motamedi
On Mon, Aug 31, 2009 at 6:28 AM, Matt Riddell 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 g
Sorry . I meant "subscriber" .
On Mon, Aug 31, 2009 at 6:31 AM, Paul Hales wrote:
> Matt Riddell wrote:
> >
> > What is a subs?
> >
> >
> A submarine. I think.
>
> PaulH
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
:
> 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 .
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> 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:H
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 wrote:
>
> I couldn't find any information on this brand of phone on the internet
> at all.
>
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
; 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 - N
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 => _XXX
ddell 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 (tha
swords, you 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 at
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)
wrote:
>
> > Please find attached my Asterisk sip.conf .
> > Can you please let me kn
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 FR3.gsm
> On Tue, Sep 8, 2009 at 1:29 PM, hadi motamedi wrote:
>
>> 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 a
age.scratchbu...@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 wrote:
>
>> No . I don't receive any error message after converting fro
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 ?
___
-- Bandwid
file.ulaw, file.alaw, file.gsm. Check
> if its 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 wrote:
>
>> Thank you . Please be informed that the *.wav files cannot be played
rmiento <
technomage.scratchbu...@gmail.com> 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 wrote:
>
>> Thank you . Please find below my original and converted sound files
>>
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 know where can I
find it ?
On Tue, Sep 8, 2009 at 2:58 PM, Steve Edwards wrote:
> On Tue, 8 Sep 2009, hadi motame
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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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 subscrib
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 wrote:
> On 09:41, Sat 26 Sep 09, hadi motamedi wrote:
> >
wrote:
> use
> Asterisk now software. You can access by IP.
>
> On Sat, Sep 26, 2009 at 2:11 PM, hadi motamedi wrote:
>
>> 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 ac
ve 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 wrote:
>>
>>> Thank you for your reply . Excuse me , you
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
___
-- B
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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aste
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 advanc
update
> then
> apt-get install gcc g++
>
> --
> Jarrod Lash,
> 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 wrote:
>
>> D
dates main contrib
"
Thank you in advance
On Sun, Nov 15, 2009 at 6:36 AM, Jarrod Lash 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 wrote:
>
>>
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 co
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 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
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 :
#whereis
On Mon, Dec 21, 2009 at 12:51 PM, Dan Journo
wrote:
> Do you have any error logs? What output do you get when you try “make
> install” with the asterisk package?
>
>
>
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>
On Tue, Dec 22, 2009 at 6:56 AM, Warren Selby wrote:
> On Mon, Dec 21, 2009 at 11:12 PM, hadi motamedi wrote:
>
>>
>> Please find below the error message that I got when issuing "make install"
>> :
>> [r...@mss-0 asterisk-1.4.26]# make install
>> ma
On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby 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 wrote:
>
>
>
> On Tue, Dec 22, 2009 at 6:56 AM, W
On Tue, Dec 22, 2009 at 9:23 AM, Tzafrir Cohen wrote:
> On Tue, Dec 22, 2009 at 07:37:50AM +0000, hadi motamedi wrote:
> > On Tue, Dec 22, 2009 at 7:28 AM, Warren Selby
> wrote:
> >
> > > And what is the output of the ./configure? Does it generate any
> errors?
mmunications Ltd.
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *hadi motamedi
> *Sent:* 22 December 2009 10:47
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users]
On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner wrote:
>
> On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote:
>
> >
> >
> >
> > On Sat, Dec 19, 2009 at 12:07 PM, Fred Posner
> wrote:
> >
> > On Dec 19, 2009, at 6:51 AM, hadi motamedi wrote:
> >
s
> not implement 'REGISTER'" problem.
>
>
> On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi wrote:
>
>>
>>
>> On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner wrote:
>>
>>>
>>> On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote:
&g
s
> not implement 'REGISTER'" problem.
>
>
> On Wed, Dec 23, 2009 at 9:12 AM, hadi motamedi wrote:
>
>>
>>
>> On Sat, Dec 19, 2009 at 1:33 PM, Fred Posner wrote:
>>
>>>
>>> On Dec 19, 2009, at 7:57 AM, hadi motamedi wrote:
&g
> 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 wrote:
>
>>
>>
>> On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham <
>> dcunnin
On Wed, Sep 9, 2009 at 4:02 AM, Jeff LaCoursiere 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 a
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
C
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 Thu, Dec 31, 2009 at 12:10 PM, Kevin P. Fleming wrote:
> 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
On Thu, Dec 31, 2009 at 6:40 AM, hadi motamedi 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
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 Mon, Jan 4, 2010 at 1:46 PM, Kevin P. Fleming wrote:
> 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 endp
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 se
On Tue, Jan 5, 2010 at 8:59 AM, Olle E. Johansson 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 I
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
allow=ala
> Now 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
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 . Ple
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 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 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 *is* obv
On Mon, Jan 11, 2010 at 6:23 AM, Zhang Shukun wrote:
> you'd better paste your dialplan snip here, in order to get specific help.
>
> 2010/1/11 Darrick Hartman :
> > On 01/10/2010 11:38 PM, hadi motamedi wrote:
> >>
> >> FWIW, he did post his question
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
--
_
On Wed, Jan 13, 2010 at 7:36 AM, Olle E. Johansson 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
&
On Wed, Jan 13, 2010 at 8:27 AM, Olle E. Johansson 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 outage yesterda
On Wed, Jan 13, 2010 at 8:49 AM, Olle E. Johansson wrote:
>
> 13 jan 2010 kl. 09.26 skrev hadi motamedi:
>
> >
> >
> > On Wed, Jan 13, 2010 at 7:36 AM, Olle E. Johansson
> wrote:
> >
> > 13 jan 2010 kl. 06.56 skrev hadi motamedi:
> >
> >
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 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
On Wed, Feb 3, 2010 at 12:17 AM, Ben Dinnerville wrote:
>
> 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
> etc), alternativ
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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New to Asterisk? Join us for a live introdu
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
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