Hi all..!! We have a trouble with asterisk 0.4.0 and audiocodes MP-108 FXO.
For someone reason some times one extension dial a number which is a PSTN call (use audiocodes for outcalling but incalling case audiocodes send this call to one extensions in asterisk like IVR), when that call was finished audiocodes don�t release the call and line FXO port but send a request for that phone number (the outcalling contest is different from incalling contest) then asterisk take that number and request to audiocodes a outcalling with that phone number, who use a other FXO port... and then a loop is showed, the worst is never release that 2 FXO ports... I don�t sure that audiocodes is whom send first request... but that stuff is random Above a brief report from CLI using show channels: Channel (Context Extension Pri ) State Appl. Data SIP/audiocodes-875b (sip_out 1 ) Up Bridged Call SIP/audiocodes-78cb SIP/audiocodes-78cb (sip 2XXXXXX 6 ) Up Dial SIP/[EMAIL PROTECTED]|300|Tr 2 active channel(s) "XXXXX" is a random phone number Where is the mistake? Can anyone help us? Sorry for my broken english :) Regards Daniel _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
