Harry, you need to start working on this yourself and stop posting to the list every half hour. Try google and keep playing until it works.
If everyone tells you the answers to everything then you don't really learn asterisk and certainly do not learn to be self sufficient and effectively troubleshoot. Thanks, Steve > -----Original Message----- > From: harry gaillac [mailto:[EMAIL PROTECTED] > Sent: Sunday, November 27, 2005 1:30 PM > To: OpenPBX.org Developers Mailing List > Subject: [Openpbx-dev] No route to destination > > Hello, > > chan_exosip2 and chansip are loaded > > -- Executing Answer("SIP/84-14da", "") in new > stack > -- Executing Dial("SIP/84-14da", "Sip/86|10") in > new stack > Nov 27 19:27:33 NOTICE[393230]: app_dial.c:1114 > dial_exec_full: Unable to create channel of type 'Sip' > (cause 3 - No route to destination) > == Everyone is busy/congested at this time (1:0/0/1) > -- Executing VoiceMail("SIP/84-14da", "b86") in > new stack > serveur1*CLI> > > harry > > > > > > > ________________________________________________________________________ __ > _ > Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger > Téléchargez cette version sur http://fr.messenger.yahoo.com > _______________________________________________ > Openpbx-dev mailing list > [email protected] > http://lists.openpbx.org/mailman/listinfo/openpbx-dev _______________________________________________ Openpbx-dev mailing list [email protected] http://lists.openpbx.org/mailman/listinfo/openpbx-dev
