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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. |
