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/00361abcdefg@voipstunt 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
Sounds like your VoIP provider is incorrectly sending you an Answer before the call actually completes. I would contact your VoIP provider. I suppose it could also be possible that YOU have an Answer() statement that is only on your VoIP trunk. I would double check that, and then contact your VoIP provider to see if they have any suggestions. Basically, SOMEONE (your or voipstunt) is answering the call before it should be answered. On 4/16/07, Adam KOSA <adamk@3a.hu> 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/00361abcdefg@voipstunt > 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070416/902dca84/attachment.htm
Yossi Ben Hagai
2007-Apr-16 07:16 UTC
[asterisk-users] duration sec and billing sec in cdr
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 <adamk@3a.hu> 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/00361abcdefg@voipstunt > 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070416/277751e1/attachment.htm
This is interesting to me.. I'm a newbie, so please forgive a dumb question, but what use is it to play a message if you don't pick up the phone first?? Who's hearing it? -----Original Message----- Adam KOSA wrote:> this is what's most likely as i have no experience in asterisk > configs. I've checked the extension.conf settins, they are: > > exten => _94./_5[05][15],1,Playback(please_wait) > exten => _94./_5[05][15],n,Set(CALLERID(name)=my_voip_username) > exten => _94./_5[05][15],n,Dial(SIP/00${EXTEN:2}@voipstunt)The Playback is your problem... you need to add |noanswer to the end of that to prevent it from automatically answering the call before it plays that recording. Trevor