search for: decarpentri

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

...05@cirpack CSeq: 2047555 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:123@62.240.245.27> Content-Length: 0 *************************************************************************** dtmfmode is RFC2833 in sip.conf Thx in advance. Regards. Guy Decarpentrie.

I'm trying to place a call from my Cisco 7960 and I'm receiving this error: Mar 1 06:19:44 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel Mar 1 06:19:58 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel I can't place calls, but I can receive them: mail*CLI> sip show channels Peer User/ANR Call ID Seq

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, We are testing a SIP solution * + ser solution for a large implementation. All the clients are nated. When a client is dialing outside the domain (to a FWD sip account for example) all is perfect ! ;-) But ,when a call is done to a sip account, the client is ringing, then the caller can hear the nated client very well, but the nated client does'nt hear anything. RTP issue no ? I've

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.