Dear all,
Anyone can point me how to soft hangup all channels using single
command ? I am using Asterisk 1.4.15.
thanks
Salaque
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Dear all
I am using AMP 1.0.10 . my voicemail system working perfectly, but now
i like to sent user PIN ( Password of the extension) number with that
mail .
how could i read the user passwd value (PIN) so that i could append
the mail format
thanks
Salaque
__
any one try that with g723 codec?
thanks
Salaque
On 8/27/06, Rosli Sukri <[EMAIL PROTECTED]> wrote:
i am also using ooh323 - it works fine on sjphone ekiga etc but i cant seem
to get it to work with ms netmeeting
On 8/26/06, atik khan < [EMAIL PROTECTED]> wrote:
> Hi,
>
> i used to work ooh32
Dear all
I am looking for suggestion , solution.
my scenario is : ppls will call to my gsm number my gsm will response
IVR from my Asetrisk box. ( with a2blling ) and could make call
now i am looking for cheap hardware solution for that gsmgateway .my
Budget is around 5000US$ for 24 gsm gate
Hello list,
I'd like to share something u all , so that i could understand whats
going on into my Asterisk box.
i have a setup like this
client(ip phone) -ip network--- [Asterisk]ip network
---[Service provider]
i have configured A2biling in my Asterisk box. so when client c
thanks for your replay,
after i disallow all codec except g723 i also confused how a2billing
is working then what i did , i removed all the codec from
/usr/lib/astersik/module without codec_g723.so .
then i saw in my log while user calling to my ivr access number a2b is
looking for gms codec as
ought to be able to tell if this would work.
Your restriction is what the service provider allows. Most (that I've used)
allow g729. I know it uses more bandwidth than g723 but nothing like G711
(ulaw or alaw) and from my experience, the quality is quite reasonable.
- Original Message
start with.
Please understand, I am trying to help and I don't know which parts (of what
I'm saying) are not entirely accurate but normally if I say something wrong
there are enough people who clamour to correct me.
- Original Message -
From: "Mohammad Salaque" <[EM
Dear all,
I have an Asterisk box running [EMAIL PROTECTED] 2.7 . and A2billing. when my
asterisk box dial using
dialcommand_param="|45|HL(%timeout%:61000:3)" its working fine .
but when i use dialcommand_param="|45|L(%timeout%)" call got drop
after 62 seconds.
i used this same setting int
Dear list
i am using Asterisk 1.2.5 with [EMAIL PROTECTED] . here is my problem.
if i dial a number (consider 79) i have to wait around 20 seconds
before my Asteisk box response. now i want to decrease this waiting
time . any idea how to do that ?
thanks
Salaque
_
ds.
You can also try pressing # after dialing the number. On most phones, that
will make it dial the number.
Good Luck,
bp
On 6/1/06, Mohammad Salaque <[EMAIL PROTECTED]> wrote:
>
Dear list
i am using Asterisk 1.2.5 with [EMAIL PROTECTED] . here is my problem.
if i dial a number
Dear List,
I am new in this world (Asterisk) and facing a problem . i want to
make some group, base on extensions .so that certain extensions could
call to certain predefine number only. let me give u all a short
example
extensions 1,2,3,4 will be group A , extensions 5,6 will be group B .
s
nto my head;
>
> Use two contextes. For example, context_A and context_B. For all group A
> extensions, make context_A their default context and group B extensions to
> context_B. Then, in each context, define only the extenions that can be
> reached.
>
> - Gabe
> ----- Orig
and one more information . user will dial to pstn number as well as
local extensions
thanks
Salaque
On 3/10/06, Mohammad Salaque <[EMAIL PROTECTED]> wrote:
> my server will be in one country . and one group will be on another
> country. . so pppoe will not work in here i thin
up
> > A extensions, make context_A their default context and group B
> > extensions to context_B. Then, in each context, define only the
> > extenions that can be reached.
> >
> > - Gabe
> > - Original Message - From: "Mohammad Salaque" <[EMAI
Dear all,
I am really new in this world, In my office i setup a Asterisk and all
extensions are working fine . i also add some trunk to route my local
calls. As some user needs to call overseas so we are looking for cheap
gateway with SIP supported.
Now . we found one gateway . they just give me
provider)
> in to sip.conf
>
> and make changes according in extension.conf for routing.
>
> ram
>
>
> On 3/22/06, Mohammad Salaque <[EMAIL PROTECTED]> wrote:
> >
> Dear all,
>
> I am really new in this world, In my office i setup a Asterisk and all
> ext
gt;
>
> exten =>
> _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
>
> just example above should help you
>
> ram
>
>
> On 3/22/06, Mohammad Salaque <[EMAIL PROTECTED]> wrote:
> > thanks ram
> >
> > all they give me an ip address. and
Dear List,
I am facing another strange problem . some of my envisions like to use
other prepaid card (whatever they found in market) but when they dial
that access number (phone number to put the pin) they get IVR (Please
provide your pin number ) but when my user press pin its not going
through,
thanks Dovid ,
i solved that yes it was DTMF issue .
thanks
Salaque
On 3/24/06, Dovid Bender <[EMAIL PROTECTED]> wrote:
> probably a DTMF issue. Try changing it. Font have the
> link here. Go to voip-info.org and search for DTMF
> type.
>
> --- Mohammad Salaque <
Hi all ,
I am gettign this warning in my asterisk log after installing g723 codec
:WARNING[3675] samples/codec_g723.c: Received a G.723.1 frame that was
4 bytes from RTP
what that mean ?
thanks
Salaque
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Hello list,
Another newbie question,. if I put "disallow=all" and "allow=g723"
my sip.cof does it mean that extension could only communicate using
g723 ?
bellow is one of my extension example
[10112]
username=10112
type=friend
secret=x
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5
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