I gave you the syntax in ael format, if you want to use extensions.conf
you'll have to use the syntax that's applicable, which is:
[start-audio]
exten => s,1,Playback(silence/1)
On 04/03/11 14:14, Olivier CALVANO wrote:
Hi Mark
Thanks for your answer, but i am new in asterisk ;=) the "context
start-audio ..."
i put it into the extension.conf ?
because i have a error:
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 136 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:17] WARNING[28078]: pbx_config.c:1499 pbx_load_config:
==!!== Unknown directive: s at line 135 -- IGNORING!!!
thanks for your help
olivier
2011/4/3 Mark Murawski<[email protected]>:
In that situation, I've had to do a pickup macro that kind of "primes" the
audio.
Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio))
context start-audio {
s => {
Playback(silence/1);
}
}
The above might help... What it does is plays an audio track on the callee's
channel (SIP/MyOperator-xxxx) before bridging the audio.
On 04/03/11 12:01, Olivier CALVANO wrote:
Hi
i use this into my extension :
exten => _00339xxxxxxxx,1,Set(foo=${SIP_HEADER(To)})
exten => _00339xxxxxxxx,2,Set(cut1=${CUT(foo,:,2)})
exten => _00339xxxxxxxx,3,Set(CLI=${CUT(cut1,>,1)})
exten => _00339xxxxxxxx,4,Set(toexten=${CUT(CLI,@,1)})
exten => _00339xxxxxxxx,5,Noop(ORIGINAL NUMBER : [ ${toexten} ])
exten => _00339xxxxxxxx,6,AGI(Ddi-Network.agi,${toexten})
exten => _00339xxxxxxxx,7,Set(CALLERPRES()=prohib_not_screened)
exten => _00339xxxxxxxx,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt)
exten => _00339xxxxxxxx,9,Hangup
and i have in sip.conf:
[MyOperator]
type=peer
host=host-of-my-operator
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=g723
defaultuser=0033xxxxxx
secret=xxxxx
When i call directly from [MyOperator], no probleme i have sound/Voice
but when a customer call to the "00339xxx..", the call are correct,
asterisk
call to my standard "SIP/MyOperator/${NUMAPPEL}" but no sound/voice
(i receive the call without problems, only sound off)
anyone have a idea of this problems ?
bye
Olivier
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