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David Norton

2004-Dec-26 14:16 UTC

[Asterisk-Users] Incoming Calls

Hi All,
 
I have the following scenario, it may already have been answered elsewhere,
but I cant find the solution.
 
I already have a PBX and would like to start implementing asterisk. I have
ordered a 4 port card from the asterisk store (2 port FXS and 2 port FXO)
and am waiting for it to arrive.
 
I do not want to plug my incoming lines into my FXO ports yet as not all the
desks have IP phones yet. I will put 2 extension lines of my existing PBX
into asterisk. When a call comes in, it will ring on all extensions,
including the asterisk ones. Here is where the problem is.
 
When the call comes in, it will be ringing on 2 of the FXO ports, and all
the other phones in the office. I would like various / all the IP phones to
ring, however asterisk must not answer the call while that is happening or
else the normal extension would not continue ringing. Obviously when an IP
phone answers it will then pick up the call and connect the 2. Is this
possible, or is this how it normally works by default?
 
Thanks in advanced
 
 
David Norton
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Rich Adamson

2004-Dec-27 06:04 UTC

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[Asterisk-Users] Incoming Calls

> Here is where the problem is.
>  
> When the call comes in, it will be ringing on 2 of the FXO ports,
> and all the other phones in the office. I would like various / all
> the IP phones to ring, however asterisk must not answer the call 
> while that is happening or else the normal extension would not 
> continue ringing. Obviously when an IP phone answers it will then 
> pick up the call and connect the 2. Is this possible, or is this
> how it normally works by default?
Maybe. Part of the answer is dependent upon exactly how your existing
pbx handles the call.

The approach I'd use for testing purposes is _not_ to ring both
extensions to asterisk, but rather just one of them. When that
extension rings, asterisk's fxo card will sense the ringing and
the logic within your dialplan will have something like:
 exten => s,1,Dial(${PHONE1}&${PHONE2}) 
that will cause two sip phones to ring. You can add more sip phones
to that statement if you'd like. If one of those sip phones answers 
the call, the fxo port will go off-hook (to your existing pbx), 
causing it to believe the call was answered; the existing pbx analog
phones should then stop ringing.

If an existing pbx analog extension answers the call, ringing to the
asterisk fxo port will stop, and therefore ringing to the sip phones
will stop a few seconds later.

There will likely be a lag of time between ringing of analog phones
and ringing of sip phones (by one or two rings), which might be
somewhat disturbing to people that can hear both phones ringing.
Should someone answer an analog extension first and someone answers
a ringing sip phone seconds later, the sip phone user will hear
nothing more then dialtone (depending upon how much lag actually
exists). 

The above essentially says that one of the existing pbx to asterisk
fxo interfaces must be dedicated to your special ringing arrangement.

C F

2004-Dec-28 11:22 UTC

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[Asterisk-Users] Incoming Calls

---------- Forwarded message ----------
From: C F <shmaltz@gmail.com>
Date: Tue, 28 Dec 2004 13:21:45 -0500
Subject: Re: [Asterisk-Users] Incoming Calls
To: Rich Adamson <radamson@routers.com>


I didn't try Dial but I did try wait and it didn't help. I'll try
dial
and see what happens. It might take a while until I do that, since I'm
waiting for a new TDM400 (the other one I installed by a client).


On Tue, 28 Dec 2004 10:32:23 -0600, Rich Adamson <radamson@routers.com>
wrote:> I'd have to guess that in your example, the exten=s entries are the
> root of the issue and is answering the call when you didn't expect it.
> Try something like this:
>  [default]
>  exten => s,1,Dial(SIP/3010)
>  exten => s,2,Hangup
> where 3010 is a valid extension. You should find that zap/4 is not
> answered until you pick up exten 3010.
>
> Also, based only on what you're showing below, it does not look like
> the contexts are working the way that you think they should. It
> appears the [default] either drops through and executes the statements
> in [incoming], or, there is something else going on in your specific
> case where the contexts aren't what you expect.
>
> ------------------------
> > I don't know what I did wrong but it didn't work. Here is how
I
> > configured it (i have a TDM400, configured with 4 fxo, channel 4 was
> > the one I wanted to share):
> > zapata.conf
> > .......
> > context=incoming
> > channel=1-3
> > context=default
> > channel=4
> > ==========> > extensions.conf
> > .......
> > [default]
> >
> > [incoming]
> > exten=s,1,do some code
> > =====================> > I left the default context blank with
no extensions b/c I didn't want
> > it to pick up. However * would pick up and in the console I get:
> > invalid extension s,1 in context default, then it would look for the t
> > extension. But it picked up after 2 rings. what am I doing wrong?
> >
> >
> >
> > On Tue, 28 Dec 2004 05:31:12 -0600, Rich Adamson
<radamson@routers.com> wrote:
> > > Not sure why it didn't work for you unless we are talking
about two
> > > different things. It does work for me and has been working just
fine
> > > for over a year now.
> > >
> > > ------------------------
> > > > Just a note on this. I tried using an external device with
the TDM400
> > > > configured as 4 FXO to ring even with asterisk. But no
matter how I
> > > > configured it, asterisk always picked up. and the external
device
> > > > didn't ring (just the first ring for CallerID to come
in).
> > > >
> > > >
> > > > > > Here is where the problem is.
> > > > > >
> > > > > > When the call comes in, it will be ringing on 2 of
the FXO ports,
> > > > > > and all the other phones in the office. I would
like various / all
> > > > > > the IP phones to ring, however asterisk must not
answer the call
> > > > > > while that is happening or else the normal
extension would not
> > > > > > continue ringing. Obviously when an IP phone
answers it will then
> > > > > > pick up the call and connect the 2. Is this
possible, or is this
> > > > > > how it normally works by default?
> > > > >
> > > > > Maybe. Part of the answer is dependent upon exactly how
your existing
> > > > > pbx handles the call.
> > > > >
> > > > > The approach I'd use for testing purposes is _not_
to ring both
> > > > > extensions to asterisk, but rather just one of them.
When that
> > > > > extension rings, asterisk's fxo card will sense the
ringing and
> > > > > the logic within your dialplan will have something
like:
> > > > >  exten => s,1,Dial(${PHONE1}&${PHONE2})
> > > > > that will cause two sip phones to ring. You can add
more sip phones
> > > > > to that statement if you'd like. If one of those
sip phones answers
> > > > > the call, the fxo port will go off-hook (to your
existing pbx),
> > > > > causing it to believe the call was answered; the
existing pbx analog
> > > > > phones should then stop ringing.
> > > > >
> > > > > If an existing pbx analog extension answers the call,
ringing to the
> > > > > asterisk fxo port will stop, and therefore ringing to
the sip phones
> > > > > will stop a few seconds later.
> > > > >
> > > > > There will likely be a lag of time between ringing of
analog phones
> > > > > and ringing of sip phones (by one or two rings), which
might be
> > > > > somewhat disturbing to people that can hear both phones
ringing.
> > > > > Should someone answer an analog extension first and
someone answers
> > > > > a ringing sip phone seconds later, the sip phone user
will hear
> > > > > nothing more then dialtone (depending upon how much lag
actually
> > > > > exists).
> > > > >
> > > > > The above essentially says that one of the existing pbx
to asterisk
> > > > > fxo interfaces must be dedicated to your special
ringing arrangement.
> > > > >
> > > > > _______________________________________________
> > > > > Asterisk-Users mailing list
> > > > > Asterisk-Users@lists.digium.com
> > > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > > To UNSUBSCRIBE or update options visit:
> > > > >   
http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > >
> > > > _______________________________________________
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users@lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > To UNSUBSCRIBE or update options visit:
> > > >    http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > > ---------------End of Original Message-----------------
> > >
> > >
>
> ---------------End of Original Message-----------------
>
>

Martin Roy

2005-Feb-02 13:07 UTC

head link

[Asterisk-Users] Incoming calls

OK I have 12 phone lines connected to 3 digium TDM04B cards on the same 
server. I must do the following thing :

The first 10 lines will be use by one company and the 2 left by another 
one. For outgoing calls it's quite easy I just create 2 different group 
and let them dial on a different one. But for incoming calls how can I 
setup asterisk to answer on the first 10 lines with one message and on 
line 11 and 12 with another one?

If I put the s,1, Answer thing it will answer all 12 lines with the same 
message...

I'm sure it's easy but I just don't know how to do it.

Thanks

Martin

Richards, Jim

2005-Feb-02 13:21 UTC

head link

[Asterisk-Users] Incoming calls

Martin,

  You would want to put them into different contexts (in the zapata.conf).
Using  different contexts you can slice and dice up your channels to your
hearts content.

  You would then be able to have an S,1,Answewr....  in that context.

Jim

-----Original Message-----
From: Martin Roy [mailto:mroy@mac.com]
Sent: Wednesday, February 02, 2005 2:08 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Incoming calls


OK I have 12 phone lines connected to 3 digium TDM04B cards on the same 
server. I must do the following thing :

The first 10 lines will be use by one company and the 2 left by another 
one. For outgoing calls it's quite easy I just create 2 different group 
and let them dial on a different one. But for incoming calls how can I 
setup asterisk to answer on the first 10 lines with one message and on 
line 11 and 12 with another one?

If I put the s,1, Answer thing it will answer all 12 lines with the same 
message...

I'm sure it's easy but I just don't know how to do it.

Thanks

Martin
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Martin Roy

2005-Feb-02 14:39 UTC

head link

[Asterisk-Users] RE: Incoming calls

So if I understand well this should do the trick : (be aware that 
context first and second include all my extensions that I haven't 
included in this and in my SIP phones use context firstinternal and 
secondinternal)

zapata.conf :

context=firstincoming
switchtype=national
signalling=fxs_ks
echotraining=800
echocancel=128
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0
txgain=0
immediate=no
busydetect=no
callprogress-no
musiconhold=default
usecallerid=yes
callerid=asreceived
group=1
channel => 1-10

context=secondincoming
switchtype=national
signalling=fxs_ks
echotraining=800
echocancel=128
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0
txgain=0
immediate=no
busydetect=no
callprogress=no
musiconhold=default
usercallerid=yes
callerid=asreceived
group=2
channel =>11-12


extensions.conf :

[firstincoming]
include => first
exten => s,1,Answer
exten => s,2,DigitTimeout(10)
exten => s,3,ResponseTimeout(20)
exten => s,4,Wait(1)
exten => s,5,Background(Intro_first)
exten => t,1,Hangup

[secondincoming]
include => second
exten => s,1,Answer
exten => s,2,DigitTimeout(10)
exten => s,3,ResponseTimeout(20)
exten => s,4,Wait(1)
exten => s,5,background(Intro_second)
exten => t,1,Hangup


[firstoutgoing]
ignorepat => 9
exten => _9NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _9NXXNXXXXXX,2,Playback(invalid)
exten => _9NXXNXXXXXX,3,Hangup

[secondoutgoing]
ignorepat => 9
exten => _9NXXNXXXXXX,1,Dial(Zap/g2/${EXTEN:1})
exten => _9NXXNXXXXXX,2,Playback(invalid)
exten => _9NXXNXXXXXX,3,Hangup

[firstinternal]
include => first
include => firstoutgoing
include => second

[secondinternal]
include => second
include => secondoutgoing
include => first

Richards, Jim

2005-Feb-02 15:26 UTC

head link

[Asterisk-Users] RE: Incoming calls

That looks like what I have...

Jim Richards
Computer Security Officer
Wisconsin Dept of Transportation

-----Original Message-----
From: Martin Roy [mailto:mroy@mac.com]
Sent: Wednesday, February 02, 2005 3:39 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RE: Incoming calls


So if I understand well this should do the trick : (be aware that 
context first and second include all my extensions that I haven't 
included in this and in my SIP phones use context firstinternal and 
secondinternal)

zapata.conf :

context=firstincoming
switchtype=national
signalling=fxs_ks
echotraining=800
echocancel=128
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0
txgain=0
immediate=no
busydetect=no
callprogress-no
musiconhold=default
usecallerid=yes
callerid=asreceived
group=1
channel => 1-10

context=secondincoming
switchtype=national
signalling=fxs_ks
echotraining=800
echocancel=128
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0
txgain=0
immediate=no
busydetect=no
callprogress=no
musiconhold=default
usercallerid=yes
callerid=asreceived
group=2
channel =>11-12


extensions.conf :

[firstincoming]
include => first
exten => s,1,Answer
exten => s,2,DigitTimeout(10)
exten => s,3,ResponseTimeout(20)
exten => s,4,Wait(1)
exten => s,5,Background(Intro_first)
exten => t,1,Hangup

[secondincoming]
include => second
exten => s,1,Answer
exten => s,2,DigitTimeout(10)
exten => s,3,ResponseTimeout(20)
exten => s,4,Wait(1)
exten => s,5,background(Intro_second)
exten => t,1,Hangup


[firstoutgoing]
ignorepat => 9
exten => _9NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _9NXXNXXXXXX,2,Playback(invalid)
exten => _9NXXNXXXXXX,3,Hangup

[secondoutgoing]
ignorepat => 9
exten => _9NXXNXXXXXX,1,Dial(Zap/g2/${EXTEN:1})
exten => _9NXXNXXXXXX,2,Playback(invalid)
exten => _9NXXNXXXXXX,3,Hangup

[firstinternal]
include => first
include => firstoutgoing
include => second

[secondinternal]
include => second
include => secondoutgoing
include => first

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dean collins

2005-Feb-02 15:40 UTC

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[Asterisk-Users] RE: Incoming calls

According to Volume 1 Asterisk Docs

To create an FXO channel on the same TDM400P card, we list all the
settings for the channel and then define the channel number. Instead of
only having signalling be fxo_ks though we want the signalling to be
fxs_ks. Because the other settings haven't been changed
(signalling=fxs_ks replaces the previous value of signalling), they stay
the same. Which means that while channel 1 takes on the values
language=en,context=default, switchtype=national, and signalling=fxo_ks;
channel takes on the values language=en, context=default,
switchtype=national, and signalling=fxs_ks.


You don't need to repeat all of the settings for the second group so it
should read.....


zapata.conf :

context=firstincoming
switchtype=national
signalling=fxs_ks
echotraining=800
echocancel=128
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0
txgain=0
immediate=no
busydetect=no
callprogress-no
musiconhold=default
usecallerid=yes
callerid=asreceived
group=1
channel => 1-10
group=2
channel =>11-12


Can anyone with more experience confirm this is correct? (I'm a total
newb when it comes to configs).


Cheers,
Dean





-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Richards,
Jim
Sent: Wednesday, February 02, 2005 5:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] RE: Incoming calls

That looks like what I have...

Jim Richards
Computer Security Officer
Wisconsin Dept of Transportation

-----Original Message-----
From: Martin Roy [mailto:mroy@mac.com]
Sent: Wednesday, February 02, 2005 3:39 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RE: Incoming calls


So if I understand well this should do the trick : (be aware that 
context first and second include all my extensions that I haven't 
included in this and in my SIP phones use context firstinternal and 
secondinternal)

zapata.conf :

context=firstincoming
switchtype=national
signalling=fxs_ks
echotraining=800
echocancel=128
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0
txgain=0
immediate=no
busydetect=no
callprogress-no
musiconhold=default
usecallerid=yes
callerid=asreceived
group=1
channel => 1-10

context=secondincoming
switchtype=national
signalling=fxs_ks
echotraining=800
echocancel=128
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0
txgain=0
immediate=no
busydetect=no
callprogress=no
musiconhold=default
usercallerid=yes
callerid=asreceived
group=2
channel =>11-12


extensions.conf :

[firstincoming]
include => first
exten => s,1,Answer
exten => s,2,DigitTimeout(10)
exten => s,3,ResponseTimeout(20)
exten => s,4,Wait(1)
exten => s,5,Background(Intro_first)
exten => t,1,Hangup

[secondincoming]
include => second
exten => s,1,Answer
exten => s,2,DigitTimeout(10)
exten => s,3,ResponseTimeout(20)
exten => s,4,Wait(1)
exten => s,5,background(Intro_second)
exten => t,1,Hangup


[firstoutgoing]
ignorepat => 9
exten => _9NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _9NXXNXXXXXX,2,Playback(invalid)
exten => _9NXXNXXXXXX,3,Hangup

[secondoutgoing]
ignorepat => 9
exten => _9NXXNXXXXXX,1,Dial(Zap/g2/${EXTEN:1})
exten => _9NXXNXXXXXX,2,Playback(invalid)
exten => _9NXXNXXXXXX,3,Hangup

[firstinternal]
include => first
include => firstoutgoing
include => second

[secondinternal]
include => second
include => secondoutgoing
include => first

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David Sampson

2005-Jun-06 12:25 UTC

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[Asterisk-Users] Incoming Calls

I have 2 4-port Digium FXS cards in my system.  I would like to play a
different recording based on which trunk rings.  Any pointers?

Thanks

 

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Carlos Chavez

2005-Jun-06 14:54 UTC

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[Asterisk-Users] Incoming Calls

On Mon, 2005-06-06 at 15:25 -0400, David Sampson wrote:> I have 2 4-port Digium FXS cards in my system.  I would like to play a
> different recording based on which trunk rings.  Any pointers?
> 
> Thanks
> 
>  	This is really a no brainer if you read the documentation.  Simple have
each channel go to a different context in your dialplan where you can
tell it to use a different message.

-- 
Carlos Ch?vez
Director de Tecnolog?a
Telecomunicaciones Abiertas de M?xico
Tel: +52-55-91169161 Ext 2001

Andres Tello Abrego

2005-Jul-21 23:53 UTC

head link

[Asterisk-Users] incoming calls

youa re using -v option multiple times at startup.
That message is perfectly fine.



ali kia wrote:> 
> 
> hi ;
> 
>  
> 
> our  * handle good  the outgoing calls but 4 incaming calls we have this 
> msg :
> 
> 
> Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 
> (Ring/Answered)...
> 
> someone have an idea ??,
> 
>  
> 
> thx in advance,
> 
> 
> 
>
<http://secure.caramail.lycos.fr/services/content/advdetail.jsp?advid=advprotekon&advsvc=advsecure&TARGETCODE=FR_footermail_link>
> CaraMail met en oeuvre un nouveau *Concept de S?curit? Globale* 
>
<http://secure.caramail.lycos.fr/services/content/advdetail.jsp?advid=advprotekon&advsvc=advsecure&TARGETCODE=FR_footermail_link>
> ? partir de 1,49 euros par mois
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

salahssaid2.salah

2005-Jul-22 03:07 UTC

head link

[Asterisk-Users] incoming calls

<html><head><style type="text/css">body{font:12px
Arial;margin:3px;overflow-y:auto;overflow-x:auto}p{margin:0px;}blockquote, ol,
ul{margin-top:0px;margin-bottom:0px;}</style></head>

<body><div style="DISPLAY: block; FONT-SIZE: 12px; FONT-FAMILY:
Arial"><BR>
<P>hi ;</P>
<P>&nbsp;</P>
<P>our&nbsp; * handle good&nbsp; the outgoing calls but 4 incaming
calls we have this msg&nbsp;:</P>
<P><BR>Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got
event 2 (Ring/Answered)...<BR><BR>someone have an idea ??,</P>
<P>&nbsp;</P>
<P>thx in advance,<BR></P>
<P>
<HR color=#dee7ee>
</P>
<P>&nbsp;<BR></P></br><p
style="margin-top:11px;padding-top:3px;background-image:
url(http://mail.lycos.co.uk/Images/Mail/_content/dot.gif);background-repeat:
repeat-x;background-position: 0px 0px;"><a
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CaraMail met en oeuvre un nouveau <a
href="http://secure.caramail.lycos.fr/services/content/advdetail.jsp?advid=advprotekon&advsvc=advsecure&TARGETCODE=FR_footermail_link"
target="_blank"><font
color="#CC0000"><b>Concept de Sécurité
Globale</b></font></a> à partir de 1,49 euros par
mois</div></body></html>

ali kia

2005-Jul-22 03:59 UTC

head link

[Asterisk-Users] incoming calls

<html><head><style type="text/css">body{font:12px
Arial;margin:3px;overflow-y:auto;overflow-x:auto}p{margin:0px;}blockquote, ol,
ul{margin-top:0px;margin-bottom:0px;}</style></head>

<body><div style="display: block; font-family: Arial; font-size:
12px;"><br>
<br>
<p>hi ;</p>

<p>&nbsp;</p>

<p>our&nbsp; * handle good&nbsp; the outgoing calls but 4 incaming
calls we have this msg&nbsp;:</p>

<p><br>Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got
event 2 (Ring/Answered)...<br><br>someone have an idea ??,</p>

<p>&nbsp;</p>

<p>thx in advance,</p>
<br></br><p
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mois</div></body></html>

salahssaid2.salah

2005-Jul-22 09:41 UTC

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[Asterisk-Users] incoming calls

?
	thk u very match 4 the answer ;
               we have use asterisk -v as u said in startup but we have always
the same problem
                in the begining we used asterisk -vvcv.
               we like know too if the order of v v c v  is important

thanx again

> De: Andres Tello Abrego <criptos@aullox.com>
> A: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
> Objet: Re: [Asterisk-Users] incoming calls
> Date: Fri, 22 Jul 2005 06:53:19 +0000
> youa re using -v option multiple times at startup.
> That message is perfectly fine.
> 
> 
> 
> ali kia wrote:
> > 
> > 
> > hi ;
> > 
> >  
> > 
> > our  * handle good  the outgoing calls but 4 incaming calls we have
this
> > msg :
> > 
> > 
> > Jul 21 11:13:05 NOTICE[12067]: chan_zap.c:5405 ss_thread: Got event 2 
> > (Ring/Answered)...
> > 
> > someone have an idea ??,
> > 
> >  
> > 
> > thx in advance,
> > 
> > 
> > 
> > <<a
href=http://secure.caramail.lycos.fr/>http://secure.caramail.lycos.fr/</a>/services/content/advdetail.jsp?advid=advprotekon&advsvc=advsecure&TARGETCODE=FR_footermail_link>
> > CaraMail met en oeuvre un nouveau *Concept de S?curit? Globale* 
> > <<a
href=http://secure.caramail.lycos.fr/>http://secure.caramail.lycos.fr/</a>/services/content/advdetail.jsp?advid=advprotekon&advsvc=advsecure&TARGETCODE=FR_footermail_link>
> > ? partir de 1,49 euros par mois
> > 
> > 
> >
------------------------------------------------------------------------
> > 
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > <a
href=http://lists.digium.com/mailman/listinfo/asterisk-users>http://lists.digium.com/mailman/listinfo/asterisk-users</a>>
To UNSUBSCRIBE or update options visit:
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> Asterisk-Users@lists.digium.com
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Protek-on: CaraMail met en oeuvre un nouveau Concept de S?curit? Globale -
www.caramail.com

Tzafrir Cohen

2005-Jul-22 11:45 UTC

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[Asterisk-Users] incoming calls

On Fri, Jul 22, 2005 at 04:41:01PM +0000, salahssaid2.salah wrote:
> > From: Andres Tello Abrego <criptos@aullox.com>
> > Date: Fri, 22 Jul 2005 06:53:19 +0000
> 
> > youa re using -v option multiple times at startup.
> > That message is perfectly fine.
And thus see quite a few messages that are not necessarily errors.
> thk u very match 4 the answer ;
> we have use asterisk -v as u said in startup but we have always the 
> same problem in the begining we used asterisk -vvcv.
> we like know too if the order of v v c v  is important
> 
No. The order is not important. 

-- 
Tzafrir Cohen         | tzafrir@jbr.cohens.org.il | VIM is
http://tzafrir.org.il |                           | a Mutt's  
tzafrir@cohens.org.il |                           |  best
ICQ# 16849755         |                           | friend

FaberK

2005-Oct-06 04:57 UTC

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[Asterisk-Users] Incoming calls

Hi,
stupid question:
how can I let to call an extensions from outside?
Untill now, I've just the possibility to call our number and then,
after the system answer, dial the extension.
My sistem is like this:
SER -> internal extensions
Asterisk -> incoming/outgoing gateway.

FaberK
--
.:FaberK:.

Josh

2006-Mar-15 07:54 UTC

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[Asterisk-Users] Incoming calls

Hi,

I run an asterisk server. The configuration is very basic.
Here is my question :
When someone calls my phone line, which is connected to an FXO card,
asterisk is answering using the context :

; Incoming calls goes to this default context  :
[incoming-rtc]
include => postes-sip
;
exten => s,1,Goto(menu,1)
exten => s,2,Hangup
;
exten => menu,1,SetVar(count=0)
exten => menu,2,Answer
exten => menu,3,Background(silence/1)
exten => menu,4,Background(josh/welcome-msg)
exten => menu,5,Background(silence/5)
exten => menu,6,SetVar(count=$[${count} + 1])
exten => menu,7,GotoIf($[${count} < 1]?4) ; Repeat 3 times
exten => menu,8,Goto(s,2)

When a friend calls, I would like for him to enter a 4 digit password
in order to access to a sub-menu, if  no password is entered, then the
welcome msg is said ...

Any hints on how to do that ??

Thanks a lot !

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