Unable to create channel of type 'Sip' Try changing "Sip" for "SIP" in your dial-plan (extensions.conf is a file found in /etc/asterisk/ by default).
;) Ariel ----- Original Message ----- From: "harry gaillac" <[EMAIL PROTECTED]> To: "OpenPBX.org Developers Mailing List" <[email protected]> Sent: Sunday, November 27, 2005 3:30 PM 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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.7/182 - Release Date: 11/24/2005 _______________________________________________ Openpbx-dev mailing list [email protected] http://lists.openpbx.org/mailman/listinfo/openpbx-dev
