When i try to call '36946811' from the outside the call gets through, but is rejected and the sound file is not played, this is my conf and sip debug output:
## sip.conf [general] context=incoming register => 36946811:[EMAIL PROTECTED]/1234 port=5060 bindaddr=0.0.0.0 srvlookup=yes ## extensions.conf [incoming] exten => 36946811,1,Background(hello-world) ## sip debug *CLI> <--- SIP read from 87.54.25.114:5060 ---> INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: <sip:87.54.25.114;ftag=688c7f1d;lr=on> Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK70b1.34ff9f57.0 Via: SIP/2.0/UDP 192.168.2.5:5060;received=62.107.1.48;branch=z9hG4bK-d8754z-c59c0dcc191940e4-1---d8754z-;rport=5060 Max-Forwards: 16 Contact: <sip:[EMAIL PROTECTED]:5060> To: <sip:[EMAIL PROTECTED]> From: "Harry"<sip:[EMAIL PROTECTED]>;transport=UDP;tag=688c7f1d Call-ID: NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Type: application/sdp User-Agent: Zoiper rev.417 Content-Length: 311 v=0 o=Z 0 0 IN IP4 192.168.2.5 s=Z c=IN IP4 192.168.2.5 t=0 0 m=audio 8000 RTP/AVP 3 110 97 8 0 101 a=fmtp:97 mode=30 a=fmtp:101 0-15 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=direction:active <-------------> --- (14 headers 15 lines) --- Sending to 87.54.25.114 : 5060 (no NAT) Using INVITE request as basis request - NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ. Found peer 'musimi' <--- Reliably Transmitting (NAT) to 87.54.25.114:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK70b1.34ff9f57.0;received=87.54.25.114 Via: SIP/2.0/UDP 192.168.2.5:5060;received=62.107.1.48;branch=z9hG4bK-d8754z-c59c0dcc191940e4-1---d8754z-;rport=5060 From: "Harry"<sip:[EMAIL PROTECTED]>;transport=UDP;tag=688c7f1d To: <sip:[EMAIL PROTECTED]>;tag=as0f99b309 Call-ID: NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35c07307" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ.' in 32000 ms (Method: INVITE) <--- SIP read from 87.54.25.114:5060 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 87.54.25.114;branch=z9hG4bK70b1.34ff9f57.0 From: "Harry"<sip:[EMAIL PROTECTED]>;transport=UDP;tag=688c7f1d Call-ID: NTRmMjg4ZGFlZDU2MDUxOTAxZjQ0OGNiYTFiOTMxOGQ. To: <sip:[EMAIL PROTECTED]>;tag=as0f99b309 CSeq: 2 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Reliably Transmitting (NAT) to 62.107.1.48:5060: OPTIONS sip:[EMAIL PROTECTED]:5060;rinstance=d815b062f3a40a5e SIP/2.0 Via: SIP/2.0/UDP 67.207.147.205:5060;branch=z9hG4bK1945b111;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as00c1a604 To: <sip:[EMAIL PROTECTED]:5060;rinstance=d815b062f3a40a5e> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 26 Apr 2008 14:46:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 62.107.1.48:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 67.207.147.205:5060;branch=z9hG4bK1945b111;rport=5060 Contact: <sip:192.168.2.5:5060> To: <sip:[EMAIL PROTECTED]:5060;rinstance=d815b062f3a40a5e>;tag=d4846e53 From: "asterisk"<sip:[EMAIL PROTECTED]>;tag=as00c1a604 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS User-Agent: Zoiper rev.417 Allow-Events: message-summary Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS _______________________________________________ -- 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