Hi,

 

 I downloaded the stable branch of asterisk a couple of month ago, and I'm using it as a SIP UA voicemail server with SER, and my setup works fine.

 I do have a list of phones defined in voicemail.conf, in the sip.conf file I only have the setup of asterisk as a peer registering to ser. The extensions.conf file contain the extensions that link to the voicemail application. This setup is working as expected so far.

 

 I downloaded the latest cvs yesterday, and with the same config files nothing work anymore: asterisk denies a caller to leave a voicemail with Forbidden-403 code as if the caller needed to authenticate with asterisk, instead of just establishing an rtp session with it and just act as a SIP UA.

 Has anything changed recently in regards to have asterisk acting not as a sip server but just as a sip ua ?

 

This is a snippet of what I have in the extensions.conf file:

 

exten => _1959XX,1,Answer

exten => _1959XX,2,Wait(1)

exten => _1959XX,3,Voicemail(u${EXTEN})

exten => _1959XX,4,Wait(1)

exten => _1959XX,5,Hangup

 

 

 

 Thanks.

 

 Samy.

 

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