On 8/10/06, Olivier Jacques <[EMAIL PROTECTED]> wrote: > Hi, > > this is probably because in your scenario you don't use the [call_id] > keyword. > Please look at how the UAC scenario is done (./sipp -sd uac |less). > > Olivier. > > > On 8/10/06, Juan Antonio Alvarez <[EMAIL PROTECTED]> wrote: > > > Hi, I was wondering if this was the expected behaviour. > I thought that callid was supposed to change on every new INVITE. But > this is not what happens. Every new invite has the same callid so I > think Asterisk handles it as it was a redirection from a previous > message. > > Am I right? > > Thanks, > > Juan >
Just for the record: This happened for a while but then I just coldn't reproduce it. Most probably was something I was doing wrong. I'll keep you informed if I see it again. Thanks and sorry, Juan ------------------------------------------------------------------------- Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 _______________________________________________ Sipp-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/sipp-users
