Displaying 20 results from an estimated 28 matches for "extesions".
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extensions
2005 May 20
5
Newbie on IVR
Hi,
I get fascinated when I dial someone and get an IVR play " for accounts department press 1, for sales, press 2 or hold the line for an operator" and then have MOH play before the line is picked up at the desired extesion.
Please, permit me as I know this will be one of the dumbest questions to ask in a group like this. I'll apprecaite any specific guide/instruction.
Thanks
2005 May 31
4
Extension context question
I have a very simple question .
I have 2 internal extension 301 and 300 sip phone . I want to these extesion can call each other, and ext 300 can call outside to pstn, and ext 301 to call internatonal.
How can I do that ?
[x1]
exten => 300,1,Dial(SIP/300)
include => pstnlocal
[x2]
exten => 301,1,Dial(SIP/301)
include =>international
[pstnlocal]
exten =>
2004 Jul 12
1
R: How to make * don't strip the leading 0
> Is it possible to tell asterisk not to strip the leading 0
> of *incoming* MSNs? I use asterisk with i4l and whenever
> I get a call from an long-distance party, the leading 0, which
> should be there according the german numbering, is not.
Are you *really* sure that the 0 is transmitted in the CLI, and that it isn't stripped already by the phone company? I think the easiest
2008 Mar 19
1
fxo tdm400p issue
hi, all
I have configure tdm400p analog fxo card.
that's ok.
but how to chek that is working properly or not.
i chek with ztcfg -vvvv and zttool .
that's ok.
i want to dial from my fxo port to another extesion.
zaptel.conf
------------------
fxsls=1,2,3,4
defaultzone=in
loadzone=in
zapata.conf
----------------
context=mycontext
signalling=fxl_ls
group=1
channel=1-4
thanks' in
2005 Mar 31
1
firefox-1.0.2-1.4.1.centos4 unstable?
Has anyone else noticed that firefox is less stable after the recent
security update?
I'm getting several occurrences per day of "just goes poof," i.e.
probably seg fault.
--
Collins
When I saw the Iraqi people voting three weeks ago, 8 million of them,
it was the start of a new Arab world.... The Berlin Wall has fallen.
- Lebanese Druze leader Walid
2010 Apr 27
2
Connect 2 asterisks servers
Hi!
I need some help
Well i have this cenario:
1 ip04 running asterisk [A]
1 pc running asterisk [B]
I nedd to make calls from A to B, and B to A. Via sip
The A-B calls are working. Now I need to configure the dial plan to call B-A
either to sip numbers and Fxs.
Anyone can help me?
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2006 May 29
2
Asterisk Internal sip calls I can´t send/recive
When i made internal call into my LAN using x-lite sip phone client I
retrive in askterisk CLI :
-----------
ERROR
----------
Verbosity is at least 6
-- Remote UNIX connection
-- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack
-- Called 201
May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call
2004 Apr 22
3
How to get call back when transfer fails
I searched the 22490 messages I have in my own personal asterisk-users archive
and have not found the answer, and it also does not appear on the wiki.
I have a SIP phone and a regular phone on a TDM400P FXS interface. Extensions
are 100 and 101, respectively.
On the SIP phone I can hit #, get the "Transfer" prompt and enter an extension
I want to transfer to. No problem. I can do
2009 Jul 03
2
Trigger an action when B number answers the call
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2003 Jun 11
0
All extensions busy
Hi
Firstly could I thnk everyone who has helped me so far,
I just have a couple of queries
I have not had chance to debug this much yet
but When using the tdm40p all extesions busy themselves out, and * cannot rint
the extensions for incoming calls
is this because I don't have a hangup statement at the end of the incoming
context? if not has anyone any idea?
does anyone have a quick and dirty IAX confiuration sample
Thanks in advance
Robb
2005 Feb 03
0
key in number after 'h' extension
Hi, asterisk gurus:
My purpose is to key in some number after the call is
finished. The number keyed in will be stored in the
database with the phone number dialed. But whenever a
key is pressed in/after h extesion, asterisk exits the
call flow.
Does anybody has a solution for this? Is DEADAGI is
possible solution?
Really need help! Thanks!Thanks!Thanks!
Manny
2005 May 24
0
silence in virtual extension
hi, I'm with problems:
1- I'm a brazilian and dont speak english hehehehehe
2- heheh my asterisk dont speak in micprohone in virtual extension. while extension is local comunication is perfect, but while a extesion is virtual is possible just listen.
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2005 May 29
0
Custom Extension on AMP
I've been using AMP to manage my * test system. I've been trying to
activate an extension that I don't want AMP to manage. It would appear
that the extesion definitions are placed in the appropriate "custom"
files which are then added with an include command to the appropriate
master file (sip.conf,extension.conf, etc). So far I've not been able
to get it to work. Anyone
2005 Jul 14
0
Plzzzzz tell me how to register users in oh323.conf
...meeting or anysoftphone )
how can i call any extension .... first of all tell me how to register a uid n password there .... let's say i've a user
type=friend
username=adeel
secret=adeel
context=incoming
mailbox=31
plz plz plzzzzzz send me just a sample oh323.conf n related portion of extesions.conf
containg above info ....
---------------------------------
Start your day with Yahoo! - make it your home page
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2005 Jul 17
0
oh323.conf ... how to regitster users ... tell me PLZZZZZZ
...meeting or anysoftphone )
how can i call any extension .... first of all tell me how to register a uid n password there .... let's say i've a user
type=friend
username=adeel
secret=adeel
context=incoming
mailbox=31
plz plz plzzzzzz send me just a sample oh323.conf n related portion of extesions.conf
containg above info ....
---------------------------------
Start your day with Yahoo! - make it your home page
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2006 Mar 10
0
Voice Mail woe
Hi
i have installed AAH 2.6 and configured some extensions
the calls are working fine. but if i dont answer a call then
it says " the person at extension " and hangs up .
it doesnt spell out the extesion number nor it goes to voice mail box.
*************************** Asterisk CLI log ****************************
dialparties.agi: Extension 200 is available...skipping checks
--
2007 Jul 04
2
Call still in queue after Reject Signal
Hi,
I have a queue with maxlen=1, and when i make a call, the call enters
into the queue,
but he doesn't exit from it after a reject signal received from the
agent??
please, have you any idea how to remove calls after a reject signal???????
Thanks.
Rachid
2011 Jun 08
1
CallerID issue
Hi List,
I am making outgoing call from asterisk to GSM network with the help of VoIP
trunk(SIP trunk) then I am not geting any caller ID at destination end. Is
this the asterisk issue or VoIP trunk issue?
Is this is due to asterisk then how we solve it? I already user
Set(CALLERID(num)=XXXXXXXXXXX) in dialplan.
-----
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
2007 Jan 29
2
UFS (Unix File System) support
Hi,
I need to mount a UFS (Unix File System) partition from an external
device (ie, live cd).
I've seen the kernel-2.6.x-x.x.x.xxx.plus.c4 has UFS support and I've
several questions about:
?How I know what FS are supported by my kernel?
?Is CentOS LiveCD edition builded with UFS support?
?Is there another way to get UFS support without the complete
installation of a new kernel?
Thanks.
2006 Jun 28
1
Help with incoming SIP routing
...is I will grab any incoming SIP call form the IP
address of my SIP trunk that matches XXX-XXX-XX00 and pass it to my
XXX-XXX-XX00 context in extensions.conf
[XXXXXXXX00]
type=friend
defaultip=69.67.248.51
host=69.67.248.51
fromuser=XXXXXXXX00
nat=no
context=XXXXXXXX00
insecure=very
And a look at extesions.conf:
// // My thought is here I will route my incoming calls to a DID i haven't
specifically routed to my default context (GoTo(XXX))
[sip-default-in]
exten => s,1,Answer()
exten => s,2,Playback(beep)
exten => s,2,Ringing
exten => s,3,Wait,1
exten => s,4,GoTo(XXX)
// // My th...