Hi, I have a new asterisk box running asterisk 1.2.24 on open suse 10.3 on an acer aspire motherboard. It has a TDM card with 3 fxos and 1 FXS, where an incoming line is plugged and also analog phone plugged to the FXS port. Am faced with the problems below.
- For conversations between analog phone and sip phone, Analog phone can't here the SIP user but Sip user hears. - Calling the PSTN from the Analog phone, still the analog phone can't hear but the PSTN user hears him saying "hello." repeatedly. Any help appreciated? I attempted a SIP debug and this is a sample out out: <-- SIP read from 192.168.209.1:48099: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.209.253:5060;branch=z9hG4bK661b7c81;rport=5060;received=192.168.209.253 From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as7b41af2a To: "Ananth" <sip:[EMAIL PROTECTED]>;tag=2bb81ff3969 Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY Content-Length: 0 Server: SJphone/1.65.377a (SJ Labs) ----- --- (11 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.102.10:49166 Found description format PCMU Found description format telephone-event Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: <sip:[EMAIL PROTECTED]> set_destination: Parsing <sip:[EMAIL PROTECTED]> for address/port to send to set_destination: set destination to 192.168.102.10, port 5060 Transmitting (NAT) to 192.168.209.1:48099: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.209.253:5060;branch=z9hG4bK4162221c;rport From: "analog-phone" <sip:[EMAIL PROTECTED]>;tag=as2b73e0bc To: <sip:[EMAIL PROTECTED]>;tag=2ae01fe36af Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Regards, Tim _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
