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Hi all, I've note that the variable DIALEDPEERNUMBER is broken. Now i want to know if exist another method to retreive the called number on *, and, if it's possible, an example ;) Regards.

Hi all, In order to do loadbalancing between my two *, i wanted to stock all things concerning voicemail on a NFS partition... I see that the voicemail system put his files onto two differents directories : /var/spool/asterisk/voicemail/mycontext etc. and /var/lib/asterisk/voicemail/mycontext etc. I've two questions : Why ? and how can i do to centralize the destination of the messages AND

If you have two asterisk systems how do you hook them up together so the users of one system can make calls onto the other system. Thanks Steve steve@17q.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040927/69058745/attachment.htm

Hi all, I've a problem in DTMF dialog between * and a Cicpack Transit Node Class 4-5. The call initiat by a mgcp phone pass by the cirpack and arrive in SIP on *, everything is ok (negociation and phone call) but when we try to use the voicemail, Asterisk don't understand DTMF. Here are some logs (SIP debug on) on a DTMF '2' receive :

Hi all, How is it possible to do loop with * ? I want to redirect ALL calls initiate by a SIP channel on itself without 'treatment' by muy * box. Regards.