This is the "full" log that I get after my trial run: Aug 24 14:15:51 VERBOSE[3710] logger.c: -- Registered SIP '202' at 192.168.1.250 port 9810 expires 120 Aug 24 14:15:52 VERBOSE[3710] logger.c: -- Registered SIP '201' at 192.168.1.251 port 8529 expires 120 Aug 24 14:15:55 NOTICE[3710] chan_sip.c: Peer '202' is now UNREACHABLE! Last qualify: 0 Aug 24 14:15:56 NOTICE[3710] chan_sip.c: Peer '201' is now UNREACHABLE! Last qualify: 0 Aug 24 14:16:07 DEBUG[3710] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Aug 24 14:16:07 DEBUG[3710] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Aug 24 14:16:10 DEBUG[3710] chan_sip.c: Setting NAT on RTP to 0 Aug 24 14:16:10 WARNING[3710] chan_sip.c: Invalid host in c= line, 'IN IPV4 192.168.1.251' Aug 24 14:16:10 DEBUG[3710] chan_sip.c: SIP message could not be handled, bad request: b475318241b3dca93128681e6f079093 192.168.1.251
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Friday, August 24, 2007 10:41 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Can't create audio conversation between softphonesthrough Asterisk Hello, I have two user machines, each with a jain-sip-applet-phone installed on it. I use the following process to try to make a call: 1. Register each phone with the Asterisk server (working). 2. Add a contact in each phone which is the other user. (Get a "489 Bad Event" SIP error shown below in red) [EMAIL PROTECTED] has been added to your contacts. null send request: SUBSCRIBE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE From: <sip:[EMAIL PROTECTED]>;tag=8505 To: <sip:[EMAIL PROTECTED]> Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585 Max-Forwards: 2 Contact: <sip:[EMAIL PROTECTED]:8386;transport=udp> Content-Length: 0 <message from="192.168.1.251:8386" to="192.168.1.10:5060" time="1187721756281" isSender="true" transactionId="z9hg4bk361290cad5885dbc4a03b5951cc85585" callId="[EMAIL PROTECTED]" firstLine="SUBSCRIBE sip:[EMAIL PROTECTED];transport=udp SIP/2.0" debugLine="0" > <![CDATA[SUBSCRIBE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE From: <sip:[EMAIL PROTECTED]>;tag=8505 To: <sip:[EMAIL PROTECTED]> Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585 Max-Forwards: 2 Contact: <sip:[EMAIL PROTECTED]:8386;transport=udp> Content-Length: 0 ]]> </message> <message from="192.168.1.10:5060" to="192.168.1.251:8386" time="1187721756281" isSender="false" statusMessage="normal processing" transactionId="z9hg4bk361290cad5885dbc4a03b5951cc85585" firstLine="SIP/2.0 489 Bad Event" callId="[EMAIL PROTECTED]" debugLine="0" > <![CDATA[SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK361290cad5885dbc4a03b5951cc85585;received=192.168.1.251 From: <sip:[EMAIL PROTECTED]>;tag=8505 To: <sip:[EMAIL PROTECTED]>;tag=as2cf724e9 Call-ID: [EMAIL PROTECTED] CSeq: 1 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY Content-Length: 0 3. Try to call that contact to create an audio conversation.(Get a "488 Not Acceptable Here" SIP error shown below in blue) Get chat session: [EMAIL PROTECTED] Chat Session added: [EMAIL PROTECTED]:gov.nist.applet.phone.ua.gui.ChatFrame[frame0,0,0,750x430,invalid,hidden,layout=java.awt.BorderLayout,title=In conversation with [EMAIL PROTECTED],resizable,normal,defaultCloseOperation=DISPOSE_ON_CLOSE,rootPane=javax.swing.JRootPane[,4,23,104x1,invalid,layout=javax.swing.JRootPane$RootLayout,alignmentX=0.0,alignmentY=0.0,border=,flags=16777673,maximumSize=,minimumSize=,preferredSize=],rootPaneCheckingEnabled=true] 5 4 3 0 send request: INVITE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: <sip:[EMAIL PROTECTED]>;tag=2085 To: <sip:[EMAIL PROTECTED]> Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2 Max-Forwards: 2 Contact: <sip:[EMAIL PROTECTED]:8386;transport=udp> Content-Type: application/sdp Content-Length: 114 v=0 o=201 908031 909400 IN IP4 192.168.1.251 s=- c=IN IPV4 192.168.1.251 t=0 0 m=audio 2448 RTP/AVP 5 4 3 0 <message from="192.168.1.251:8386" to="192.168.1.10:5060" time="1187721758593" isSender="true" transactionId="z9hg4bk583c7ea7c1a8da8576583356f821c9c2" callId="[EMAIL PROTECTED]" firstLine="INVITE sip:[EMAIL PROTECTED];transport=udp SIP/2.0" debugLine="0" > <![CDATA[INVITE sip:[EMAIL PROTECTED];transport=udp SIP/2.0 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE From: <sip:[EMAIL PROTECTED]>;tag=2085 To: <sip:[EMAIL PROTECTED]> Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2 Max-Forwards: 2 Contact: <sip:[EMAIL PROTECTED]:8386;transport=udp> Content-Type: application/sdp Content-Length: 114 ]]> </message> <message from="192.168.1.10:5060" to="192.168.1.251:8386" time="1187721758609" isSender="false" statusMessage="normal processing" transactionId="z9hg4bk583c7ea7c1a8da8576583356f821c9c2" firstLine="SIP/2.0 488 Not acceptable here" callId="[EMAIL PROTECTED]" debugLine="0" > <![CDATA[SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.1.251:8386;branch=z9hG4bK583c7ea7c1a8da8576583356f821c9c2;received=192.168.1.251 From: <sip:[EMAIL PROTECTED]>;tag=2085 To: <sip:[EMAIL PROTECTED]>;tag=as2f851644 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY Content-Length: 0 Has anyone ever tried using these Jain-sip-applet-phones and got them to work? I have read up on these errors, and it looks like the 489 error doesn't like the SUBSCRIBE request, while the 488 error doesn't seem to accept the INVITE request made. I am not sure if this is a problem with Asterisk, incompatibility between Asterisk and the phones, or just the phones. Any thoughts that may help me resolve these issues would be greatly appreciated. Thanks very much, Denis _______________________________________________ --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 _______________________________________________ --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
