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:[EMAIL PROTECTED]
.: <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.
_______________________________________________
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
