On Mon, 2009-11-30 at 13:31 +0000, Nitin wrote: > Scott Lawrence <scott.lawrence <at> nortel.com> writes: > > If you'd like help with working out the issues, I suggest starting by > > choosing just one problem, carefully describing your complete > > configuration, and collecting traces of the network traffic for that one > > problem and posting them here. > > > > See: > > > > http://sipx- > wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer#Getting_SIP_ > Messages_to_display > > > > > > I did see the sipviewer but we donot have any linux boxes around with X > installed. Can you give any link for windows based sip viewer?
It was on that page: http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer#Standalone > Or maybe guide me how to use wireshark to collect sip packets. > > I will summarize problems > > a) Vegastream or GXW4104(as gateway) can only transfer calls to AA or Polycom > phones. They cannot transfer calls to Cisco. > > b) I can call cisco-cisco or Polycom-cisco or Cisco-Polycom. I can call Cisco- > AA. > > c) I cannot transfer call Cisco-Cisco. I can transfer Cisco-Polycom, but not > polycom-Cisco. > > > Before everybody gets confused, Cisco phones cannot recv calls from/to AA or > gateway. They are regsitering well becasue I can call from other phones to > Cisco > (internal) > > > And also, for Cisco - dialplan.xml is a requirement (on wiki it says > otherwise) > becasue if dialplan.xml is not present, the phone will dial only 1 digit. Support for configuring Cisco phones is not strong, but that should not be causing transfer failures. Pick one of those cases (the simplest would probably be AA-Cisco) and collect trace data using sipx-trace. _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
