Displaying 20 results from an estimated 1000 matches similar to: "DISA problem in 1.8.0"
2004 Dec 23
2
DISA restart from begining
Hi,
Is there a way to restart the DISA to the enter phone number? For
instance, Bell Calling Cards let you hit # at any point which lets you
enter another number to call. This is useful to reduce the number of
digits dialed and to utilize per-minute calls.
I was not able to find anything on the web.
Thanks,
-Ryan
2004 Jun 11
2
extensions question
ser forwards a sip message with extension 99999996 to asterisk which
plays my 'userisoffline' message and hangs up and should stop here but
instead asterisk continues to process the match everything extension ._
and dials out which is not what I want...
if I change the starting priority of the Dial app to a higher level
than 3 asterisk stops after the hangup but then doesn't accept
2005 Jan 11
1
Dial Out Errors
Hey, I'm having some errors whenever I dial out and I can't dial in at
all. I'm using NuFone as my provider just so you know.
Jan 11 17:39:46 WARNING[1771]: chan_oss.c:413 soundcard_setinput:
Unable to re-open DSP device: No such device
Jan 11 17:39:46 WARNING[1771]: chan_oss.c:572 oss_write: Unable to set
device to input mode
Jan 11 17:39:46 WARNING[1771]: app_dial.c:359
2004 Apr 25
2
asterisk dials wrong numbers ?!?
Hi,
I've got an important question:
I use an E100P directly connected to PSTN, but it does not *really* work as it should
be:
exten => 1000,1,Dial(Zap/1/1234)
BUT: It does NOT dial "1234" but it says in debug mode:
-- Called 1/72976451
Apr 26 00:53:00 WARNING[10251]: chan_zap.c:5979 zt_pri_error: PRI: !! Facility
message shorter than 14 bytes
-- Channel 1, span 1 got
2006 Feb 09
4
Problem win Unicall
I am having a strange problem with an asterisk servier using R2 Unicall
in Mexico. Most calls go through fine but some of them give me an error like
this:
-- Executing Dial("SIP/86-db41", "Unicall/g2/014448343600") in new stack
-- Called g2/014448343600
Feb 9 21:44:39 WARNING[23069]: chan_unicall.c:2644 handle_uc_event: Unicall/2
event Dialing
Feb 9 21:44:45
2007 Sep 13
2
DTMF error on asterisk
Dear all
I have asterisk 1.4.11 on centos 4.x i have installed 2 PRI on is asterisk and it is working fine but i got this DTMF error on asterisk CLI what is it ??
-- Zap/36-1 is ringing
-- Zap/36-1 answered SIP/5406-9fa59770
-- Channel 0/1, span 2 got hangup request, cause 31
[Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: Unable to forward voice or
2009 Nov 05
1
Asterisk 1.4 DISA is jumoing after one digit in the DISA context
Dear list,
I have problems with DISA on an specific server with Asterisk 1.4.26.2.
After starting DISA I can only press one key and DISA is jumping direct
into the context without waiting for further digits.
In dtmf.log I found this:
[Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin '7' received on
SIP/214-00d92db0
[Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin passthrough
2003 Dec 24
8
G729 troubles
Hello,
I've successfully installed Asterisk from last CVS and configured it
for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip
server.
All are work fine at G711 codecs, but then I disable all codecs except
g729 some calls failed (Not all calls. Some calls passed at g729
succesfully).
All my devices configred to use only g729 and I don't see other codecs
at mgcp or sip
2006 Nov 28
1
Attn: DISA Experts(Strange problem with DISA)
Hi Friends,
I am facing a strange problem with DISA. I have installed and configured Trixbox. I've created a secret extension i.e., 555 and called this extension in Digital Receptionist using custom extension i.e., created in extensions_custom.conf file.
When I call from my mobile phone to my PSTN number, which is connected to FXO port, my IVR is responding. After entering my DISA
2005 Mar 21
1
DISA Hangs up after DTMF is sent
Hey, this is happening to anyone who I try this with. We get into the
DISA, then hear the dial tone. Dial 1 then start dialing the number,
and it hangs up. I thought adding a wait time after the DISA may help,
I was wrong. Here is what I have thus far in the DISA extentions.
[DISA]
exten => 7,1,DISA(no-password||"Scheda" <565> 455-1337)
exten => 7,2,Wait(45)
exten =>
2005 Jul 25
2
DISA disconnects
DISA is currently disconnecting when I dial 8888 to access DISA.
Below is my extensions.conf file from A@H and some lines which shows
the disconnect. Should DISA be loaded as a module in modules.conf?
When I do a 'show applications' i see that DISA is there. Help!
--------------------------------------
;Asterisk CLI as I placed a call from cell into the system.
Playing
2005 Mar 19
1
DISA -> macro = congestion
When I use DISA I get congestion when I try to reach 1-800-number:
Here is the context:
[disa]
exten => 087,1,Answer
exten => 087,2,DigitTimeout,8
exten => 087,3,ResponseTimeout,20
exten => 087,4,Authenticate(985)
exten => 087,5,DISA(951|disa-access)
[disa-access]
include => tollfree
include => outgoing-voipjet
[tollfree]
;
; terminate toll-free no.'s via fwdnet
; US
2005 Jun 05
2
Disa - how it returns on user not dialing any numbers ?
Hi,
I'd like to use DISA properly for my case - I'd like to handle it right, if
user when in DISA doesn't dial any number - how does Asterisk return from
DISA cmd ?
I'd like to dial some default number if user doesn't dial anything or give
him some message - but I don't know what gets executed after DISA if nothing
is dialed ....
I'm reading this on wiki, but
2008 Jan 04
1
Unable to forward call on SIP channel after SIP response 302 Moved Temporarily
Hi,
I have the following problem that when asterisk receives SIP response 302 it
cannot forward the call
I get such debug:
[Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel
type registered for 'Local'
[Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to
create local channel for call forward to 'Local/poczta at routing-sip' (cause =
66)
2005 May 10
2
DISA
We are using DISA with local SIP users. The user enters in a 2 digit
code then they get a dialtone and the phone dials out. The problem is
that the calls waits 10 seconds after the outgoing number is dialed, no
matter what I put for the timeout values. Anyone else using DISA that
has run into this?
exten => _2X,1,Answer
exten => _2X,2,DigitTimeout(2)
exten =>
2005 Aug 16
1
DISA over Zap (TE110P) issues on * STABLE 1.0.9
Hi !
Did anyone had issues/managed to solve issues with DISA over Zap channels on
* 1.0.X (STABLE) ?
I have a situatuion where DTMFs that should be recognized in DISA work over
SIP channels and do not work over ZAP channels (Zap channels are on TE110P)
I have in default context:
exten=> 299,1,DISA(no-password|default)
and I have SIP extension 200 in [default] and I have Zap trunk which
2006 Apr 08
2
question about DISA
Lists,
?
? Hi, good day, i was being task to create a DISA access for internal
purpose of the company, i'm having a problem to work with it with
authentication, but i think it's really a straight forward thing to do,
can someone enlight me on this. thanks
?
sample code snippet
?
???? exten => 5,Goto(inward,s,1)
?
[inward]
?
?????????? exten => s,1,Disa(1234|outgoing)
?????????? ;
2003 Oct 14
1
DISA and ringing tone
Hi
I am using DISA to get my Polycom SoundPoint400 with H323 firmware to
connect to *
I have it working, but when I dial SIP end points there is no ringing tone
on the phone. DISA gives dial tone but does not give ringing (if I
understand correctly it is because it expects to transmit sound created by
terminating side of the call)
Is there a way to make DISA application to generate ringing
2009 Sep 02
2
DISA() fails to recognize dtmf where WaitExten() succeeds (DAHDI-PRI)
Is there any known reason that the DISA() routine should behave
differently than WaitExten() as far as recognizing DTMF tones? If
not, I suspect there's a bug here.
Try it yourself--two DID's on our PRI, numbers below let you test each routine:
It is my observation that some setups/phones DO and some DO NOT
express this variance.
--I could not show any variance on a sprint mobile phone
2004 Jul 02
0
DISA and AGI: authenticate by caller ID? (resolved)
Here is some code to do authentication by caller ID for DISA through AGI.
My original code had a bug in the Mysql query code, and there was a hangup
in the wrong place
[that's what I get for coding something at 2:00am], but the attached code
works correctly.
Take note of the REGEXP for the CallerID variable. When I tested the code
from the PSTN
it worked because there was no name component,