Displaying 20 results from an estimated 2000 matches similar to: "Hangup problem"
2005 Oct 11
2
IAX or IAX2 ?
Hi,
I have read the wiki entries on IAX(2), but I'm afraid, it still have some
questions:
I have a working connection between two Asterisk-Servers (Asterisk
1.0.7-BRIstuffed-0.2.0-RC7k on Debian 3.1) via IAX.
Does this connection work with IAX or IAX2?
When I try to load chan_iax2.so, I get the error message
chan_iax2.so]Oct 11 10:09:52 WARNING[2288]: loader.c:258
ast_load_resource:
2005 Oct 15
1
Problem with '#' key recognition
Hi,
I seem to be unable to get Asterisk to recognise the '#' key being
pressed to acknowledge an incoming call from a queue. No matter how many
times I press the key to acknowledge, the Asterisk server acts as if I
have not.
I have installed the ztdummy module, and it seems that Asterisk is
picking it up OK (mainly since it's not moaning about permissions for
IAX timing any
2006 Apr 05
2
chan_modem_i4l delay
Hi,
I currently use? Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian sarge with a kernel 2.4.27 on a P4 3Gig with 1Gig
of memory
When i use i4l on any call, the called party ( on the telco operator side ) ear me with a delay of 1 sec after 1
minutes , 2 sec after 3 minutes and so on...
After a quart hour, the delay make the conversation just
2005 May 30
1
I865, HFC-S etc.
Hi,
I'am having some problems with new mainboards and 3xHFC-S cards.
The the first problem was with interrupts, I mean if HFC-S card was using
interrupt i.e. 21 or higher - it didn't work.
Solved by disabling APIC.
However, still the driver behaves a little bit strange.
If card 0 & 1 is TE and 2 is NT, TE works fine, but NT is not working at
all.
If card 0 is NT and 1 & 2 TE - all
2004 May 18
1
VoiceMailMain dumps user back into my incoming context after leaving a message
I have a dial plan that includes a company phone directory as a main menu
option. If they just sit at the main menu, after 20 seconds, they are
transferred to the operator. If the user picks an extension from the
directory, they are transferred to the proper extension. If the called
number is not available, they are transferred into VoiceMailMain. They
leave a message, and hang up. The hang
2006 May 03
1
Running applications when a queued call is answered
Hello,
I'm experimenting with Asterisk for possible use in a call center.
I'm trying to figure out how to run applications when an agent answers
a call in the queue. I see that the queue itself supports a very
limited range of applications; for example, I can give a URL to the
Queue() application to SendURL(), or an announcement to read to the
agent. I'd like to do some slightly
2007 May 17
2
Blacklist
Hello All,
I was wondering where does Asterisk stores the blacklist numbers?
I looked into the dialplan and it shows that it
*"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB?
hyperion*CLI> show dialplan app-blacklist-add
[ Context 'app-blacklist-add' created by 'pbx_config' ]
'1' => 1.
2005 Mar 21
2
Ext matching problems
Hello everyone...
I'm trying to get up a testing pbx installation. Following instructions
of what've read from the handbook and from asterisk's wiki, I wrote the
dial plan as follows:
[general]
;
;
static = yes
;[globals]
;
[default]
;
exten => 0,1,Answer()
exten => 0,2,Playback(fcopba1)
exten => 0,3,Hangup()
exten => *0,1,Answer()
exten => *0,2,Record(fcopba1:gsm)
2003 Jul 07
1
Dial plan doesn't seem to save properly
When I first to the "add extension" the "show dialplan" has the lines that
say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and
restart the lines with "SIP/" are gone.
************************
"Show dialplan" before:
************************
asterisk01*CLI>
[ Context 'default' created by
2005 Jun 14
2
# no longer working
Hi list,
For months everything worked super here in our setup.
This week I implemented some new idea in our webbased
calendar system. I thought it would be nice to have an
option that tells asterisk you are not available for calls
during an appointment.
For this to work I could no longer use the ringgroup setup:
Dial(SIP/10&SIP/11&SIP/12,40,tr)
So I thought, why not use the Local channel
2008 Jun 25
1
included context not being prioritized properly
I have an "outbound-ld" context as follows:
[ Context 'outbound-ld' created by 'pbx_config' ]
'_1NXXNXXXXXX' => 1. Macro(enumdial|${EXTEN}) [pbx_config]
102. Wait(1) [pbx_config]
103. Set(LINE=${IF($[${LINE}=pots]?link2voip:${LINE})}) [pbx_config]
2005 Feb 24
2
Delay after entering digits with IVR
I have a [start] context that all my inbound and '0' calls are routed
into.
Because of the way I want to set my system up, I want to prompt the user
to enter a 1 if they know the extension, or a 2 for a directory and
nothing else.
It works, however there is a 5 to 10 second delay after enter the 1 or 2
before the system responds.
I have read over the wiki on how asterisk handles digit
2006 Feb 19
3
Cisco 7905 can't register
My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k
on Debian stable). It could, however, register with another
installation of Asterisk and the settings on the phone (apart from the
SIP proxy address) haven't changed since then.
On the new Asterisk box my sip.conf contains this:
[jeremy]
type=friend
regexten=801
allow=g729
host=dynamic
secret=PASSWORD
nat=yes
2005 Aug 18
1
Newbie Trying to make 'catch all extension' but is catching voicemail exit!
Greetings,
Running CVS HEAD about 3 weeks old,
I have been beating my head trying to get this to work properly..
Or at least figure out what's going on.
Maybe I have done things wrong...
I have created a 'catch all' extension at the end of our last context
where all phones & voicemail extension exist.
This catch all is included in all and works quite nicely except
when voicemail
2007 Oct 15
2
Voicemail issues in 1.4.11
Asterisk isn't playing my voicemail greetings even though they are defined. Below are the relevant configs(from show dialplan) as well as the level 3 verbose messages asterisk is giving. Also a listing of the directory.
Asterisk just plays the "The person at extension..." message, not the greetings I have recorded.
Thanks
--
asterisk*CLI> show dialplan macro-stdexten
[
2007 Mar 29
2
L options in Dial() dont seem to work....
Hello Asterisk users,
Can someone thwack me with a clue stick please?
I am following the Asterisk TFOT book Dial() example trying to get the limit
and announcements to work as per below.
These settings seem to have no effect.
There are no warning messages after 4 minutes or every 30 secs thereafter
and the call lasts longer than 5 minutes.
gunner*CLI> show dialplan
[ Context
2010 Aug 05
1
Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
Hello.
I have been beating my head over this problem for about 6 hours now.
I have a SIP peer, who I register to (successfully), who should be
directing all incoming calls at my [default] stanza in my
extensions.conf:
[ Context 'default' created by 'pbx_config' ]
's' => 1. Wait(1) [pbx_config]
2.
2015 Dec 02
2
Issues with Twilio number incoming call and context matching
Yes, I have tried that too (i.e, exten => +17775551212,1,Log(WARNING,
TWILIO)). It does not work and NO error message in CLI.
I have also tried
http://orourketech.com/elastix-plus-sign-caller-id-messing-things/ since I
first emailed this group, but that does not seem to work either.
Here is my log:
[Dec 2 15:09:28] NOTICE[26464]: res_pjsip_session.c:1920 new_invite: Call
from
2010 Dec 20
2
Unexpected dialplan match
I was wondering why *foo at default should match '_*[0-9a-zA-Z].*0.'
in 1.6.13. Who is making the parse error, * or me?
CLI> dialplan show *foo at default
'_*[0-9a-zA-Z].*0.' =>
1. NoOp(${EXTEN}) [pbx_config]
2. Set(accountcode=${CUT(EXTEN,*,2)}) [pbx_config]
3. Set(extension=${CUT(EXTEN,*,3)}) [pbx_config]
2005 Sep 09
4
Huge Echo
asterisk-users-bounces@lists.digium.com wrote:
> In the following setup:
> call coming from a pstn line -> into FXO card -> asterisk -> SIP
> phone
>
> i get an incredible loud echo in the SIP phone (about 0,5-1s)
> (everything i speak into SIP phone microphone i hear in its
> speaker). The person calling from PSTN is not getting any echo.
Make sure you're not