Not using the CDR for billing, but I do use it to see usage and to know if it's cheaper to purchase a provider with unlimited incoming and pay-per-minute outgoing. I disabled 'SIP Transformation' in the SonicWall and so far so good (10/10 calls worked, more testing to be had, stay tuned.)
On Sat, Nov 8, 2008 at 5:12 AM, Steve Totaro <[EMAIL PROTECTED]> wrote: > Usually, calls terminating at 30 seconds is a sure sign that you need to add > an Answer() in your dialplan. Try dropping that in before you dial out. I > have seen this so many times and Answer() has always fixed the issue. The > magic number is 30 seconds. > > Depending on if you use your CDRs for anything, especially billing, you may > need to figure a way around that, since even if a call rings out, the CDR > will reflect Answered. > > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > > > > On Fri, Nov 7, 2008 at 10:38 PM, Grey Man <[EMAIL PROTECTED]> wrote: >> >> To get to the bottom of it I'd recommend determining why the ACKs are >> not getting through to Asterisk rather than trying to work around it. >> I'm actually suprised Asterisk terminates the call by default when it >> doesn't get the ACK to it's 200 Ok response that must be new for >> 1.4.22 as I haven't seen that behaviour in earlier versions. In my >> opinion it's unwarranted behaviour, if Asterisk is getting RTP then it >> should leave the call up irrespective of whether it gets an ACK or >> not. >> >> From the original SIP trace the ACK does not appear to be arriving at >> your Asterisk server at all. Try doing a packet trace on the network >> segment where the calling SIP agent is and see where it's trying to >> send the ACK to. My guess would be your firewall is incorrectly >> handling the SIP messages. Generally it's very bad news to use an ALG >> or firewall to mangle SIP packets as they almost always get it wrong. >> >> In your case there is a Record-Route header in the response so the ACK >> request should be being sent to that address. Perhaps your firewall is >> not correctly mangling that to allow the request to find its way back >> to your Asterisk server. >> >> Record-Route: <sip:216.82.224.202;lr;ftag=as3ed791f3> >> >> Regards, >> >> Greyman. >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
