Looks okay to me. either the number you are testing with your VoIP provider has an automated response which answers the call at the same sec you sent the Invite request or the provider is sending False Answer Supervision...do a sip debug and check while you make the call.
On 4/16/07, Adam KOSA <[EMAIL PROTECTED]> wrote:
Hi guys, i've installed asterisk to handle multiple voip accounts. I've looked at CDR configs, and managed to have cdr-csv files growing after each call. It would be easier to check my locak asterisk cdr's than logging into each account and check them at the provider website. i found that if i ring my sip softphone from my ata, bill seconds are counted correctly. however, if i call via a voip provider, bill seconds are counted incorrectly. Example: this call went to a pstn number New call from 551 ---> 94361abcdefg (context: internal) Dialed : SIP/[EMAIL PROTECTED] Call start: 2007-04-14 20:10:55 Answered : 2007-04-14 20:10:55 Call end : 2007-04-14 20:11:10 Duration : 15 sec Bill : 15 sec this call went to my ata from the sip softphone: New call from 551 ---> 505 (context: internal) Dialed : SIP/505|45 Call start: 2007-04-15 07:58:11 Answered : 2007-04-15 07:58:15 Call end : 2007-04-15 07:58:43 Duration : 32 sec Bill : 28 sec i've searched and google'd the wiki, but found only accounting software and cdr extensions for providers, but that's not what i need. thanks for any help Adam _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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