Hello, openser does just routing job in this case. If the sip requests reach the endpoints properly, then the issue is probably in on one of them.
As I can get from your summary, Asterisk does not have active the call to be replaced. Cheers, Daniel On 11/27/08 10:03, muhammad akl wrote: > I have the following scenario : > > > Pstn Number(1234567) <-----------> Asterisk GW <----------------> > Openser | <-------------->11803 > > | > > > | > > > | <--------------> 11801 > > Firstly extension 11803 will call the pstn number and this works fine > without no problem , after that 11803 will put 1234567 on hold and > will call 11801 , then 11803 will transfer 1234567 to 11801 (<---- the > problem now started ), what is happening now is that both 123456 and > 11801 will be on hold with 11803 after the transfer is done > > I've traced the full dialog between the three extensions and found an > interesting part , which a NOTIFY message came from asterisk and > contains this sentence : SIP/2.0 481 Call leg/transaction does not exist > > The addresses of the devices as follows : > > Asterisk Gw : 192.168.200.202 <http://192.168.200.202/> > > OpenSER : 192.168.200.10 <http://192.168.200.10/> > > 11803 : 192.168.200.222 <http://192.168.200.222/> > > 11801 : 192.168.200.224 <http://192.168.200.224/> > > > The full trace : > > http://muhammad.akl.googlepages.com/debug.txt > > Regards > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > [email protected] > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com _______________________________________________ Users mailing list [email protected] http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
