Displaying 5 results from an estimated 5 matches for "callerintros".
2007 Jun 25
2
two channels, each dropping into the same context, different behavior.
...Zap/4-1",
"SIP/lesnet_peer/12183486879|30|ptT") in new stack
-- Privacy-- callerid is empty
-- <Zap/4-1> Playing 'priv-recordintro' (language 'en')
-- <Zap/4-1> Playing 'beep' (language 'en')
-- x=0, open writing: priv-callerintros/NOCALLERID_2Zap=4-1 format: gsm,
0x81d9920
-- Recording automatically stopped after a silence of 2 seconds
-- <Zap/4-1> Playing 'auth-thankyou' (language 'en')
-- <Zap/4-1> Playing 'vm-dialout' (language 'en')
-- Called lesnet_peer/12...
2004 Jan 08
0
Re: Sound Card help -- Solution -- and a question
...undcard in the server. While I was at it, I also use it to play
the name of
the intended person being called, as around here, several can share the
same phone.
I just use the plain old sound app "play" to do this by:
exten => s,1,SystemCID(/usr/bin/play
/var/lib/asterisk/sounds/priv-callerintros/%s.gsm&)
If a sound file corresponding to the CID exists in the priv-callerintros
dir, it sounds off. Otherwise,
no intro.
Making an app to force anyone who doesn't have an entry, to record one
for them,
shouldn't be that big a deal. Right now, I have such a beast in the
privacy co...
2006 Apr 21
1
1.2.7.1 on FC5 won't make install
The make seems to go okay.
[root@somebox asterisk-1.2.7.1]# uname -a
Linux somebox.org 2.6.16-1.2080_FC5smp #1 SMP i686 i686 i386 GNU/Linux
mkdir -p /var/lib/asterisk/sounds/digits
mkdir -p /var/lib/asterisk/sounds/priv-callerintros
for x in sounds/digits/*.gsm; do \
if grep -q "^%`basename $x`%" sounds.txt; then \
install -m 644 $x /var/lib/asterisk/sounds/digits ; \
else \
echo "No description for $x"; \
exit 1; \
fi; \
done
No descri...
2007 Apr 01
5
[MACRO-SCREEN] and MACRO_RESULT
I am following the example at http://www.voip-info.org/wiki/view/Asterisk+tips+findme but I find that no matter what, the call is connected. Can anyone confirm that config is working for them? Any suggestions appreciated.
I need to transfer calls to a list of cell phones, ring all of them, allow them to screen the call, connect the call to the first number that accepts the call, and allow
2006 Nov 22
1
DTMF detection during Call
Hi
I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by
outbound SIP.
Now i want to detect DTMF-Tone Code coming from the called party to
trigger a signal.
Can this be done with asterisk? I read that the codec with DTMF
detection are ulaw and alaw. But i couldn't find a command to detect
dtmf's within a normal call.
thanks and mani greetings
Christian