Roberto Piola
2004-Aug-05 14:37 UTC
[Asterisk-Users] Strange message, and one-way audio between sip and H.323
we are trying to use asterisk for converting SIP to H.323 calls. asterisk (0.9.1) runs on the same linux (Redhat 8) box of our gatekeeper (gnugk version 2.0.8). the calls are going out through a cisco gateway. when I make a call from a SIP phone to a PSTN number reachable through the cisco gateway: asterisk diaplays Aug 5 23:24:26 WARNING[1255648560]: chan_oh323.c:2898 alerted_h323_connection: Call ip$localhost/22666 in unexpected state (PLAYONLY). I hear (on the SIP phone) clearly what the other person is saying, but the other person (on the PSTN side) hears nothing from me. gatekeeper and the cisco gateway work fine, when using H.323 terminals. The result does not change if I use a IAX phone instead of a SIP one. The gatekeeper configuration contains [RoutedMode] GKRouted=1 H245Routed=1 and has no Proxy Section I tried also [RoutedMode] GKRouted=0 H245Routed=0 and I also tried to enable the [Proxy] function on the gatekeeper, but the result is the same I tried to search the internet for the message, but I got no results Roberto Piola, Ph.D. Senior Network Engineer Divisione VAIPS ----------------------------- SOFTPEOPLE - IHNET .: Strada del Drosso 128/6 - 10135 Torino .: tel. +39 011 3473520 - mob. +39 335 6961505 - fax. +39 011 3473522 .: mail:roberto.piola@softpeople.ihnet.it .: <http://www.softpeople.it> .: <http://www.ihnet.it> Business Unit di SOFTPEOPLE ----------------------------- Questo messaggio ? destinato alle sole persone indicate e pu? contenere informazioni riservate. Se ricevuto per errore, si prega di avvisare immediatamente il mittente e cancellare l'originale. Ogni altro uso del messaggio ? vietato.