We are using Asterisk, inside an Hybrid network (LAN and WAN) and several VoIP SIP equipments like or components , PDA SoftPhone, and WiFi phone, gateway with Cisco Router (2600 and 2801) and PABX like Alcatel, Siemens, and analog phones. When We use channel_h323 to establish VoIP session between SIP Users agent and H.323 endpoints, through Gatekeeper and Gateway, we are able to make ringing on al the phones and to connect them. When we are speaking in the SIP UA microphone, H.323 endpoint can receive correctly the voice with any codec. But when, H.323 endpoint (an analog phone behind an FXS voice card registered with Gatekeeper), We can’t hear the voice in the SIP speaker. Between SIP UA or between H.323 endpoint everything is ok. We tried to specify just one codec at all endpoint and sip UA. And when we analyze SIP log or ASN.1 log we can’t see the error.
We think that more information about h323.conf file will be usefull.
Thanks for your help