Displaying 20 results from an estimated 7000 matches similar to: "FXO not picking up; baffled"
2004 Jul 26
3
ResponseTimeout, Straight to operator?
Hi,
My client wants incoming callers who do not press a digit to go straight
to the operator. Does anyone have an idea of how this could be done?
I've looked for some examples, but I'm still not clear on it.
Here's the relevant portion of my extensions.conf:
-------
; Wait 15 seconds for an answer (pick up the local phone)
exten => s,1,Wait,2
; Answer the phone
exten =>
2004 Mar 30
5
Caller entered digits ignored during wait....
Greetings,
Below is part of the contents of my extensions.conf file.
exten => s,1,Wait,1 ; Wait a second before
answering.
exten => s,2,Answer
exten => s,3,ResponseTimeout,10 ; Set the amount of
time the user
; has to
make a selection.
exten => s,4,DigitTimeout,5
2003 Jul 24
2
Voicemail() problems - Long pause after incoming message recording ended.
I'm having the following problem:
I call into my Asterisk box (RedHat Linux 9.0, 1 Digium X100P card)
to access voicemail. After dialing the appropriate extension I get
voicemail, am presented with the user's unavailable message, and can
leave a message normally.
The problem comes when I press "#" to end the recording, at which
point I am told "Your message has been
2004 Jul 28
2
Music On Hold - not working for me...
Hi all,
I'm trying to make some simple MOH (Music On Hold) working. So far I've failed
miserably - so I turn here for help.
Basically I've been using the wiki and all the sample confs I could from there
and via google.
The queue system seems to work fine with my limited setup. Just 2 IAX2 clients
where I keep Client B busy (by making it listen to mp3 via ext. 777) but logged
into
2005 Aug 23
0
X100P Clone not picking up incoming calls. [POTS]
Hello All,
I have this strange problem,
I can dial out with my sip phone and it seems to work relatively well, but
when I call in, the line just rings and rings, I get no indication in
asterisk that it's detecting an incoming call.
The strange thing is that in ztmonitor 1 -vv the rx volume goes from around
99% when the line is idle to 0% as soon as the line starts ringing.
I'm located
2006 May 01
3
auto-dail for ZAP channel, the application gets executed before the call attended
Hi All
when I try to use auto-dial to connect to
outside phone , my applications get executed before
the caller attend the calls , this happens only when I
call outside no , ie when I use
Channel: ZAP/1/0507451111 in my sample.call file , if
I use Channel:SIP/326 , it works fine
my ?sample.call? file contains
Channel: ZAP/1/0507451111
Callerid: Asterisk
MaxRetries: 2
RetryTime: 10
2006 Dec 27
1
php agi trixbox help
I have this code which was taken from the phpagi project page along with the
following in extensions_conf and the output from the asterisk CLI. When I
call the 311 extension, I does nothing then hangs up. What am I doing
wrong??
----php code------------
#!/usr/local/bin/php -q
<?php
set_time_limit(30);
require('phpagi.php');
$agi = new AGI();
$agi->answer();
$cid =
2006 Mar 22
2
Asterisk--->>Autodialling
Hi all
I'm trying to dial out with a Digium X100P card set up
on channel Zap/1 to local number (25921163).
My call file is: (out.call)
Channel: Zap/1/25921163
Callerid: 25921163
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: outgoing
Extension: s
Priority: 1
And my extension.conf is,
[outgoing]
exten => s,1,DigitTimeout,5
exten => s,2,ResponseTimeout,10
exten => s,3,Answer
2005 Aug 31
1
problems with dialing-out with Zap
Hello Guys,
I am trying to make Asterisk do dial-out calls.
It doesn't even do test calls. It never calls. I
tested everything and i am clueless. However i can
call Asterisk and it picks up the phone and executes
the dial-plan. However, my dial-plan is supposed to
do
outbound calls.
Zap is configured correctly.
I am using a TDM400 card from Digium with 4 Fxo
ports
and i have
2004 Jan 27
1
Distinctive ring Issues
Hello all!
We have a PSTN line with four numbers calling into it. There is
distinctive ring on these lines. They are are follows:
1. standard ring
2. short ring
3. long ring
4. short ring, long ring, short ring
Based on the information I have been able to find, I have created the
following entries in my zapata.conf file, to
try and weed out some of the timings:
dring1=95,0,0
2004 Jun 23
3
help needed with read()
Hi,
Greatly appreciate if some one help me with the application read().
asterisk*CLI> show application read
asterisk*CLI>
-= Info about application 'Read' =-
[Synopsis]:
Read a variable
[Description]:
Read(variable[|filename]): Reads a '#' terminated string of digits from
the user, optionally playing a given filename first. Returns -1 on hangup
or
error and 0
2004 Aug 20
1
x100p won't answer
Hi,
I just got two digium x100p clones and installed asterisk on fedora
core 2 which took some tweaking. After getting asterisk up I installed
the zaptel stuff - then modprobed zaptel, wcfxs (for the fxo cards),
which worked fine. ztcfg is showing two channels configured, but when I
start asterisk and do show channels, i see no active channels.
zapata.conf has:
signalling = fxs_ks
2004 Aug 21
3
zaptel config
Hi,
Sorry, in my last mail I wrote "wcfxs" instead of what I actually used, "wcfxo." I just got two digium x100p clones and installed asterisk on fedora
core 2 which took some tweaking. After getting asterisk up I installed
the zaptel stuff - then modprobed zaptel, wcfxo,
which worked fine. ztcfg is showing two channels configured, but when I
start asterisk and
2004 Dec 13
1
Repost: Cisco 7960 and Asterisk...not working....
Anyone help me here? I am a newbie so be gentle ;-)......
It worked once and then I played with the configs.
I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22
I have the 7690 with a SIP iamge (Whatever latest is )
I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here
2003 Jun 18
2
== Everyone is busy at this time problem
hi,
i installed asterisk and works very well, the only problem is that
when i try to call a direct number of a company that has a normal PBX
i got this error:
to 10.8.210.153:5060
== Accepting call on 'SIP/a.sampietro-f7be' (a.sampietro)
-- Executing Goto("SIP/a.sampietro-f7be", "doisdn|BYEXTENSION|1") in new stack
-- Goto (doisdn,00115601992,1)
--
2009 Feb 24
2
Multiple SIPGate accounts.
Hi all,
I have two sipgate accounts (numbers), if I have both accounts register only
one will work for incoming calls (which is all i'm interested in). However
if I disable either account the other account will work perfectly. Am I
missing something obvious?
Thanks in advance,
Ray.
Excerpts from sip.conf -
[general]
8<---- SNIP! ---->8
Register => 1212121:aaaaaaaa at
2004 May 05
7
* & ISDN-BRI-PTP & DID & ISDN4Linux does not show incoming number
Hi,
I am using Asterisk on a DSS1 ISDN-BRI with ISDN4Linux (and a Fritz Card PnP).
The ISDN-BRI is in PTP-Mode (Point to Point "german: Anlagenanschluss")
which is enabled within I4L with "hisaxctrl fcpcipnp0 7 1".
Everything works fine except that I can not see the called number/MSN
of incoming calls within Asterisk and because of this I can not route
incoming calls
2003 Oct 19
2
The Start extension
I have my sip phones going into the context [from-sip] and would like to
play an introduction message and then have the caller enter the extension.
The message (dir-info was picked just to have something) doesn't play.
Maybe I misunderstood the "s" extension. According to what I read it is
executed everytime something enters the context. Obviously something was
misunderstood.
The
2007 Jan 09
1
Asterisk 1.2.11 - ResponseTimeout being ignored
All - this is probably a simple problem, but I've been pulling my hair
out trying to figure out what I'm doing wrong. I'm building a
*simple* IVR menu. Here it is:
[main-menu]
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout(5)
exten => s,4,ResponseTimeout(30)
exten => s,5,Background(logic-main)
exten =>
2003 Dec 14
1
Error loading modem driver
When I attempt to start asterisk with my modem setup listed it will not start
attached are the error messages i get and also the modem.conf that i am currently using. Any assistance would be greatly appreciated.
running CVS ver 12/7/03, modified only to allow the RxFax and TxFax to compile and run with it (from http://www.opencall.org)
just e-mail me privately if you need more info
Thanks in