<div class="gmail_quote">Hi,</div>
<div class="gmail_quote"><br>Could someone please help me with this?</div>
<div class="gmail_quote"><br>I have a new asterisk box running asterisk 1.2.24 on open suse 10.3 on<br>an acer aspire motherboard. It has a TDM card with 3 fxos and 1 FXS,<br>where an incoming line is plugged and also analog phone plugged to the<br>
FXS port. Am faced with the problems below.<br><br>- For conversations between analog phone and sip phone, Analog phone<br>can't here the SIP user but Sip user hears.<br>- Calling the PSTN from the Analog phone, still the analog phone<br>
can't hear but the PSTN user hears him saying "hello." repeatedly.<br><br>Any help appreciated?<br><br><br>I attempted a SIP debug and this is a sample out out:<br><br><-- SIP read from <a href="http://192.168.209.1:48099/" target="_blank">192.168.209.1:48099</a>:<br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP<br>192.168.209.253:5060;branch=z9hG4bK661b7c81;rport=5060;received=<a href="http://192.168.209.253/" target="_blank">192.168.209.253</a><br>From: "asterisk" <<a href="mailto:sip%3Aasterisk@192.168.209.253">sip:asterisk@192.168.209.253</a>>;tag=as7b41af2a<br>
To: "Ananth" <<a href="mailto:sip%3AAnanth2@192.168.102.10">sip:Ananth2@192.168.102.10</a>>;tag=2bb81ff3969<br>Call-ID: <a href="mailto:7758606e479584bf2c20a80d792841f5@192.168.209.253">7758606e479584bf2c20a80d792841f5@192.168.209.253</a><br>
CSeq: 102 NOTIFY<br>Content-Length: 0<br>Server: SJphone/1.65.377a (SJ Labs)<br><br>-----<br>--- (11 headers 12 lines) ---<br>Found RTP audio format 0<br>Found RTP audio format 101<br>Peer audio RTP is at port <a href="http://192.168.102.10:49166/" target="_blank">192.168.102.10:49166</a><br>
Found description format PCMU<br>Found description format telephone-event<br>Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x4<br>(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)<br>Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1<br>
(telephone-event), combined - 0x1 (telephone-event)<br>list_route: hop: <<a href="mailto:sip%3AAnanth2@192.168.102.10">sip:Ananth2@192.168.102.10</a>><br>set_destination: Parsing <<a href="mailto:sip%3AAnanth2@192.168.102.10">sip:Ananth2@192.168.102.10</a>> for address/port<br>
to send to<br>set_destination: set destination to <a href="http://192.168.102.10/" target="_blank">192.168.102.10</a>, port 5060<br>Transmitting (NAT) to <a href="http://192.168.209.1:48099/" target="_blank">192.168.209.1:48099</a>:<br>
ACK <a href="mailto:sip%3AAnanth2@192.168.102.10">sip:Ananth2@192.168.102.10</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.209.253:5060;branch=z9hG4bK4162221c;rport<br>From: "analog-phone" <<a href="mailto:sip%3Aasterisk@192.168.209.253">sip:asterisk@192.168.209.253</a>>;tag=as2b73e0bc<br>
To: <<a href="mailto:sip%3AAnanth2@192.168.102.10">sip:Ananth2@192.168.102.10</a>>;tag=2ae01fe36af<br>Contact: <<a href="mailto:sip%3Aasterisk@192.168.209.253">sip:asterisk@192.168.209.253</a>><br>Call-ID: <a href="mailto:09727d336de2434a2a1043436322c03b@192.168.209.253">09727d336de2434a2a1043436322c03b@192.168.209.253</a><br>
CSeq: 102 ACK<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Content-Length: 0<br><br><br>Regards,<br>Tim<br></div><br>