Displaying 16 results from an estimated 16 matches for "monkeysintro".
2011 Feb 27
1
[Dahdi 2.4.0] Flash() hangs up
Hello
I need Asterisk (1.4.39.2) to simulate a flash hook (ie. hitting the
"R" key on European handsets) so I can put a call on hold, dial a
second number, and set up a conference call.
By default, linux/include/dahdi/kernel.h sets the flashtime to 750ms,
which appears to be too long for European telcos, as they seem to
expect a line cut of about 100ms.
After editing the
2004 Apr 16
2
Newbie alert: Cannot get voicemail to answer (have scoured the web for help)
...s head, downloaded this morning.
I have attached my conf files below.
Julian.
#### extensions.conf ####
[general]
static=yes
writeprotect=yes
[default]
exten => s,1,Wait,1
exten => s,2,Answer
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,10
exten => s,5,BackGround(tt-monkeysintro)
exten => s,6,BackGround(tt-monkeys)
exten => _7XX,1,Dial(zap/1/${EXTEN}|5m)
exten => _7XX,2,Voicemail(u${EXTEN})
exten => _7XX,3,Hangup
exten => _7XX,103,Voicemail(b${EXTEN})
exten => _7XX,104,Hangup
exten => #,1,Playback(demo-thanks)
exten => #,2,Hangup
exten => t,1,...
2004 Jan 19
2
Hangup detection failed
...ing to look for
that can help prevent this? The system is running on a telenet line in
Belgium. The answer dialplan I used was:
[macro-stddial]
exten => s,1,Answer
exten => s,2,Playback(transfer)
exten => s,3,Dial(${ARG2},60)
exten => s,4,Voicemail(u${ARG1})
exten => s,5,Playback(tt-monkeysintro)
exten => s,6,Playback(vm-goodbye)
exten => s,7,Hangup
exten => s,104,Voicemail(b${ARG1})
- Kim Hendrikse
2005 Sep 14
0
oh323 and Asterisk: Calls always hang up
...213.30.225.5-0148 created and attached for inbound H.323
call 'ip$213.30.225.5:42873/1893'.
-- Executing NoOp("OH323/-----@213.30.225.5-0148", "h323 Call an
49222299663-99!") in new stack
-- Executing Playback("OH323/-----@213.30.225.5-0148", "tt-monkeysintro")
in new stack
Channel OH323/-----@213.30.225.5-0148 answered.
-- Playing 'tt-monkeysintro' (language 'en')
Call 'ip$213.30.225.5:42873/1893' cleared.
-- H.323 call 'ip$213.30.225.5:42873/1893' cleared, reason 24 (Call ended
with Q.931 cause)
Sep 14 10...
2004 May 05
3
sip.conf and SIP client host= not recognized in some cases
...aa.aaa.aaa.aaa ; Address to bind to
context = default ; Default for incoming calls
[carriera]
type=friend
host=ccc.ccc.ccc.ccc
context=inbound
[carrierb]
type=friend
host=bbb.bbb.bbb.bbb
context=inbound
/etc/asterisk/extensions.conf
[inbound]
exten => _.,1,Playback,tt-monkeysintro
[default]
exten => _.,1,Congestion
Example A:
U ccc.ccc.ccc.ccc:5060 -> aaa.aaa.aaa.aaa:5060
INVITE sip:4445552574@aaa.aaa.aaa.aaa SIP/2.0..
Via: SIP/2.0/UDP ccc.ccc.ccc.ccc:5060;branch=z9hG4bK7ab24dcc..
From: "asterisk" <sip:asterisk@ccc.ccc.ccc.ccc>;tag=as3a541e32..
To:...
2004 Sep 26
3
What about a higher level configuration language
Hi all.
I've been reading through Wi-Ki and at the extensions.conf file
description (http://www.voip-info.org/wiki-Asterisk+config+extensions.conf)
The author says this:
"One day, someone is going to write a proper scripting language for
Asterisk that can understand a simpler, easier (and more traditional)
scripting syntax. All it would need to do is translate the "high
2003 Oct 30
0
SIP error: Asked to transmit frame type 64
...m
allow=ilbc
allow=ulaw
We dial "98616" here:
exten => _9XXXX,1,Playback(transfer)
exten => _9XXXX,2,Ringing
exten => _9XXXX,3,Wait(1)
exten => _9XXXX,4,Dial(IAX2/myserv:mypw@remote-regist-server/${EXTEN:1})
exten => _9XXXX,5,Congestion
exten => _9XXXX,105,Playback(tt-monkeysintro)
exten => _9XXXX,106,Hangup
my chan_sip.c:
static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct sip_pvt *p = ast->pvt->pvt;
int res = 0;
if (frame->frametype == AST_FRAME_VOICE) {
if (!(frame->subclass & ast->nativeformats)) {
--> -->...
2010 Mar 03
0
Is this a bug?
...39;m working on making one of my applications multi-lingual and
find that I have this problem. The SayDigits and SayNumber functions in
1.4.26.2 recognize but don't process the CHANNEL(language) function. Here's
a snippet to verify.
exten => 317,1,Answer
exten => 317,n,playback(tt-monkeysintro)
exten => 317,n,Set(CHANNEL(language)=es)
exten => 317,n,Wait(2)
exten => 317,n,SayDigits(123)
exten => 317,n,SayNumber(1)
exten => 317,n,playback(vm-goodbye)
exten => 317,n,Set(CHANNEL(language)=en)
exten => 317,n,Wait(2)
exten => 317,n,SayDigits(123)
exten => 3...
2008 Mar 16
4
Telemarketer Torture....
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Anyone have the telemarketer torture prompts? I would seriously like
to revive this.....
- --
James Finstrom
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.6 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFH3I8qdloC7YyaIOoRAlAjAJ9Hp+3SS2Z8179HecWIETp4RVDzWQCeMizp
fW2JPZdYl/uxG1ziUwYnHGo=
=QPbv
-----END PGP
2011 Feb 24
2
[1.4] Still can't get it to call back
...2)
exten => s,n,Hangup
;Call script to build callfile
exten => h,1,DeadAGI(/var/tmp/callback.lua,${CID})
;Context used by callfile
[callback]
;Zaptel doesn't support call progress, so just wait 10s
exten => start,1,Wait(10)
exten => start,n,Answer()
exten => start,n,Playback(tt-monkeysintro)
exten => start,n,Hangup()
======== AGI script
#!/var/tmp/lua
--Must first empty stdin
while true do
local line = io.read()
if line == "" then break end
-- Without line below, script never ends
io.write("NOOP ",line,"\n")
end
file =...
2011 Mar 03
6
[1.4] Forcing Asterisk/Zaptel to wait until callee answers?
...1/${IPPI})
exten => 8888,n,NoOp(We never get there)
exten => 8888,n,Hangup
============
;call made through callfile: Doesn't wait for off-hook
[callbacktest]
exten => start,1,NoOp(DialStatus is ${DIALSTATUS})
exten => start,n,NoOp(${CHANNEL(state)})
exten => start,n,Playback(tt-monkeysintro)
exten => start,n,Wait(2)
exten => start,n,Hangup
============
;tried this to force callfile to wait until off-hook
[callbacktest]
exten => start,1,Set(INDEX=0)
;Down, Rsrvd, OffHook, Dialing, Ring, Ringing, Up, Busy, Dialing
;Offhook, Pre-ring, Unknown
;Tried "Up", "Off...
2007 Dec 02
2
Answering Machine Detection
...> exten => 13,n,NoOp( log the NAK acknowlegement here calling the AGI script)
> ;exten => 13,n,AGI(lax/track-laxcalls.sh,${EXTEN},${CALL_ACK})
> exten => 13,n(play13),Playback(lax/lax-important-msg-from)
> exten => 13,n,Playback(tt-weasels)
> exten => 13,n,Playback(tt-monkeysintro)
> exten => 13,n,Wait(1)
> exten => 13,n,Hangup
>
>
>
>
>
> Carlos Chavez wrote:
> > I am having a bit of a problem getting AMD to work on a new server. On
> > my regular office server it works like a charm. I am running Asterisk
> > 1.4.13, Z...
2007 Oct 15
1
Answering Machine Detection
I am having a bit of a problem getting AMD to work on a new server. On
my regular office server it works like a charm. I am running Asterisk
1.4.13, Zaptel 1.4.5.1 on both machines. Both servers run CentOS 5 and
I am using a SIP trunk to send out calls (the same one on both servers).
Here is the output of a call on my office server:
-- Attempting call on Local/0445540881644 at CC2 for
2006 Feb 13
1
Asterisk: Agent logs into queue, and there are calls in the queue, but calls don't go to agent
Here is some dialog from the Console:
-- Starting simple switch on 'Zap/13-1'
Feb 10 07:22:36 NOTICE[21105]: chan_zap.c:6063 ss_thread: Got event 18 (Ring
Begin)...
-- Executing Goto("Zap/13-1", "mainmenu|s|1") in new stack
-- Goto (mainmenu,s,1)
-- Executing BackGround("Zap/13-1", "thank-you-for-calling-poker -support")
in new stack
2004 Sep 17
8
English vs American voice files
My wife's got an appropriate Southern England (Wimbledon) accent and I'm
sure she would try her hand. Does anyone have a comprehensive list of the
words that need to be said? Matt, do you have them if your wife's done a
set for French users?
Mark, if you have the kit maybe you could chop up the file? I write a
utility to chop up and compress the wave file based on some of the C
2004 Jul 18
18
Polycom IP 500 Voicemail
Hello All,
I have some Polycom IP 500 phones that I would like to have configured
for direct dialing to our voice mail system. So far I have been unable
to get the hard button labeled Voice Mail to connect to Asterisk without
first passing through the message center prompts. I have followed all
the Admin Guide instructions regarding the phones .cfg files and using