For SIP, we developed SIPSak http://sipsak.berlios.de/. (BTW -- not all SIP servers show their IP addresses in Warning header field, so the output is not always as talkative as it should be).
-jiri At 05:47 PM 3/17/2004, David Zuzga wrote: >Is there a traceroute equivalent in the VoIP world? I would like to see the >route a call takes after it gets to the gateway. Basically showing all the >hops until it reaches it's destination or PSTN termination. > >-Dave > >_______________________________________________ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Jiri Kuthan http://iptel.org/~jiri/ _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
