Displaying 9 results from an estimated 9 matches for "dialednum".
2008 Jul 16
0
ISDN Call Droping only for outgoing
...-- Starting simple switch on
'Zap/62-1'
[May 8 17:51:55] VERBOSE[4711] logger.c: -- Channel 0/31, span 2 got
hangup, cause 17
[May 8 17:51:55] WARNING[4762] channel.c: Unexpected control subclass '5'
[May 8 17:51:58] VERBOSE[4762] logger.c: -- Executing
[07973555555 at dialednum:1] GotoIf("Zap/62-1", "0?8:") in new stack
[May 8 17:51:58] VERBOSE[4762] logger.c: -- Executing
[07973555555 at dialednum:2] NoOp("Zap/62-1", "") in new stack
[May 8 17:51:58] VERBOSE[4762] logger.c: -- Executing
[07973555555 at dialednum:3] NoOp(&...
2006 Feb 01
1
SetCDRUserField not working in A@H?
...alplan looks like this:
<snip>
exten => s,n,SetCDRUserField(${dnum})
exten => s,n,Noop(cdr user field is '${CDR(userfield)')
exten => s,n,AppendCDRUserField( ${cdn})
exten => s,n,Noop(cdr user field is '${CDR(userfield)')
exten => s,n,AppendCDRUserField( ${dialednum})
exten => s,n,Noop(cdr user field is '${CDR(userfield)')
;exten => s,n,Set(CDR(userfield)=${dnum}|${cdn}|${dialednum})
;exten => s,n,Noop(cdr user field is '${CDR(userfield)')
</snip>
I get the Noop lines logged at the CLI:
<snip>
-- Executing Set...
2004 Dec 21
10
Codec Selection
Hi,
I have 2 g729 licences - what I want to do is use g729 by default but if
I get more than 2 calls at a time, use gsm for the others.
So, I put this on all my sip providers:
disallow=all
allow=g729
allow=gsm
However, this just seems to use gsm for everything. If I comment out the
gsm line, it then uses g729.
I thought it would use the codec's in the order they are allowed - is
this
2006 Jun 19
0
Call Not Disconnecting
...d for 1,2
hrs. even they made very small calls.
i have already set rtptimeout = 60, but not
disconnecting
Here is my extentions.
[main-ext]
exten => _x.,1,AGI(main-ext.pl)
exten =>
h,1,DeadAGI(/var/lib/asterisk/agi-bin/main-stop.pl)
AGI Script:
my $dialstr = "$gwtype/$gwip/" . $dialednum .
"|350|tTL(" . ($credit_time*1000) .":7000:5000)";
$AGI->exec('Dial', $dialstr);
Could please advice me how i can prevent such kind of
issue?
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protec...
2007 Apr 03
0
Faxing issues
...d asterisk-1.4.2 installed and whenever a fax call
comes in we get this. This isn't good. Any ideas?
[New Thread -1215390800 (LWP 8504)]
-- Accepting call from 'DELETED' to '539' on channel 0/1, span 1
-- Executing [539@telco-incoming:1] Set("Zap/1-1", "DIALEDNUM=539") in
new stack
-- Executing [539@telco-incoming:2] Answer("Zap/1-1", "") in new stack
-- Executing [539@telco-incoming:3] Ringing("Zap/1-1", "") in new stack
-- Executing [539@telco-incoming:4] Wait("Zap/1-1", "4") in...
2006 Jun 23
5
Asking for phone number to dial
Does anyone know where to find an example or able to provide an example of how to do the following:
When asterisk answers a call...
Ask for number to dial...then dial that number?
I am basically dialing into the asterisk box and then wanting it to take the digits I enter and dial them on an outbound zap trunk...
I basically am just not sure how to have asterisk accept the digits and then use
2006 Nov 08
0
Warning: "Channel does not have a CDR" when doing ForkCDR
...n,AMD
exten => s,n,Noop(AMDSTATUS is '${AMDSTATUS"')
exten => s,n,GotoIf($[${AMDSTATUS} = AMD_MACHINE]?lmtc,s,1:human)
exten => s,n(human),Set(NUMTRIES=1)
exten => s,n,SetCDRUserField(${dnum})
exten => s,n,AppendCDRUserField(:${cdn})
exten => s,n,AppendCDRUserField(:${dialednum})
exten => s,n(repeat),Background(Initial-greeting)
exten => s,n,Wait(.1)
exten => s,n,Flite(${fname})
exten => s,n,Flite(${lname})
exten => s,n,Background(If-this-person-press-1-else-press-2)
exten => s,n,Set(NUMTRIES=$[${NUMTRIES}+1])
exten => s,n,GotoIf($[${NUMTRIES} < 2]...
2008 Jan 08
4
Bugs??
Good Day All,
I am facing a serious problem since I started to use asterisk. I don?t know if it is a bug or some one already solved this.
Currently I am running 120 VoIP SIP channels on my asterisk server but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI ?show channels? showing us as call UP but in real there is no call.
When asterisk restarted the hanged calls removed from
2003 Jul 09
17
caller id
Hello,
is it possible to change how are caller id on incoming call from isdn,
capi lines displayed od sip phones ? ( e.g. SNOM ) standard is
1234567@domain.net. I just want only 1234567 to be displayed. is it
possible ?
regards
Marian
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