[asterisk-dev] mute calls through a sip proxy

[Antonio Ceccatelli]

*Hi guys,

Does anyone know why when I copy this file in 
/var/spool/asterisk/outgoing/  the call is mute.  ? Any help will be 
appreciated.*

=============== /* sip.call*/
Channel: SIP/123456 at 10.244.116.49
MaxRetries: 2
RetryTime: 60
WaitTime: 60
callerid: '"magik" <299>'
extension: 299


*This is my config:*

/*============= Sip.conf*/

register => asterisk127 at sipcommander         <'sipcommander' it's our 
sip proxy >

[sipcommander]
username=asterisk127
type=friend
secret=
port=5060
nat=never
host=10.244.116.49
context=sip
canreinvite=no

/*============  Extensions.conf*/

exten => 299,1,Dial(SIP/sipcommander)



*Regards
Antonio C.*

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