Göran Törnqvist
2004-Dec-15 06:15 UTC
[Asterisk-Users] Calls arent handled by asterisk - destruction of call
Hello, I?m trying to get started with asterisk/SIP so I was trying the demo that is provided in the extensions config file, but the call isn?t ?answered? by my server when I try calling the number that I registered at my SIP provider. I?ve registered with register => John.Doe:MyPass:MyUser@my-sip-provider/1000 in sip.conf and if I use ?sip debug? I can see the call is coming in but then nothing more happens (see debug output below). Also get these error messages: Scheduling destruction of call '397F980B-4DCF11D9-9E359155-12491096@my-sip-providers-ip' in 15000 ms WARNING[4863]: chan_sip.c:706 retrans_pkt: Maximum retries exceeded on call 221201580f4bd104062df83a2437e145@MY-IP for seqno 102 (Non-critical Request) Sip.conf: [general] context=demo [my-sip-provider] type=peer fromuser=MyUser secret=MyPass fromdomain=my-sip-provider context=demo extensions.conf: [demo] ; ; All the stuff in the demo ; exten => s,1,Wait,1 ; Wait a second, just for fun exten => s,n,Answer ; Answer the line exten => s,n,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => s,n,ResponseTimeout,10 ; Set Response Timeout to 10 seconds and so on That?s all I have have I missed something? Debug output from call: 192.1.1.1=my server 0123456789=my number at SIP-provider 9999999999=the number I?m calling from 213.132.103.213, 212.112.162.50=my SIP providers IPs ========================================= Sip read: INVITE sip:s@192.1.1.1 SIP/2.0 Record-Route: <sip:213.132.103.213:5060;transport=UDP;lr=true> Via: SIP/2.0/UDP 213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa655559cda50ac3600b Record-Route: <sip:0123456789@212.112.162.50;ftag=2EBE3E60-1646;lr> Via: SIP/2.0/UDP 212.112.162.50;branch=z9hG4bK4885.ddcc862.0 Via: SIP/2.0/UDP 212.112.162.22:5060 From: <sip:9999999999@212.112.162.22>;tag=2EBE3E60-1646 To: <sip:0123456789@212.112.162.50> Date: Wed, 15 Dec 2004 10:10:11 GMT Call-ID: 56F68A14-4DB811D9-95A69155-12491096@212.112.162.22 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 1458717796-1303908825-2510524757-306778262 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 9 Remote-Party-ID: <sip:9999999999@212.112.162.22>;party=calling;screen=yes;privacy=off Timestamp: 1103105411 Contact: <sip:9999999999@212.112.162.22:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 288 v=0 o=CiscoSystemsSIP-GW-UserAgent 1486 6130 IN IP4 212.112.162.22 s=SIP Call c=IN IP4 212.112.162.22 t=0 0 m=audio 16842 RTP/AVP 18 0 101 c=IN IP4 212.112.162.22 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 24 headers, 12 lines Using latest request as basis request Sending to 213.132.103.213 : 5060 (non-NAT) Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 212.112.162.22:16842 Found description format G729 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Found peer 'wx3.se' Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa655559cda50ac3600b Via: SIP/2.0/UDP 212.112.162.50;branch=z9hG4bK4885.ddcc862.0 Via: SIP/2.0/UDP 212.112.162.22:5060 From: <sip:9999999999@212.112.162.22>;tag=2EBE3E60-1646 To: <sip:0123456789@212.112.162.50>;tag=as3c0db481 Call-ID: 56F68A14-4DB811D9-95A69155-12491096@212.112.162.22 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:s@192.1.1.1> Proxy-Authenticate: Digest realm="asterisk", nonce="59e60c89" Content-Length: 0 to 213.132.103.213:5060 Scheduling destruction of call '56F68A14-4DB811D9-95A69155-12491096@212.112.162.22' in 15000 ms Sip read: ACK sip:s@192.1.1.1 SIP/2.0 User-Agent: sapphire/1.6.2.0253 Max-Forwards: 70 Via: SIP/2.0/UDP 213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa655559cda50ac3600b To: <sip:0123456789@212.112.162.50>;tag=as3c0db481 From: <sip:9999999999@212.112.162.22>;tag=2EBE3E60-1646 Call-ID: 56F68A14-4DB811D9-95A69155-12491096@212.112.162.22 CSeq: 101 ACK Content-Length: 0 9 headers, 0 lines -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041217/fb6ef7f2/attachment.htm
test
2004-Dec-16 01:59 UTC
[Asterisk-Users] Calls arent handled by asterisk - destruction of call
Hello, I'm trying to get started with asterisk/SIP so I was trying the demo that is provided in the extensions config file, but the call isn't "answered" by my server when I try calling the number that I registered at my SIP provider. I've registered with register => John.Doe:MyPass:MyUser@my-sip-provider/1000 in sip.conf and if I use "sip debug" I can see the call is coming in but then nothing more happens (see debug output below). Also get these error messages: Scheduling destruction of call '397F980B-4DCF11D9-9E359155-12491096@my-sip-providers-ip' in 15000 ms WARNING[4863]: chan_sip.c:706 retrans_pkt: Maximum retries exceeded on call 221201580f4bd104062df83a2437e145@MY-IP for seqno 102 (Non-critical Request) Can you guys help me? Thanks :) Sip.conf: [general] context=demo [my-sip-provider] type=peer fromuser=MyUser secret=MyPass fromdomain=my-sip-provider context=demo extensions.conf: [demo] ; ; All the stuff in the demo. ; exten => s,1,Wait,1 ; Wait a second, just for fun exten => s,n,Answer ; Answer the line exten => s,n,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten => s,n,ResponseTimeout,10 ; Set Response Timeout to 10 seconds .and so on. That's all I have.have I missed something? Debug output from call: 192.1.1.1=my server 0123456789=my number at SIP-provider 9999999999=the number I'm calling from 213.132.103.213, 212.112.162.50=my SIP providers IPs ========================================= Sip read: INVITE sip:s@192.1.1.1 SIP/2.0 Record-Route: <sip:213.132.103.213:5060;transport=UDP;lr=true> Via: SIP/2.0/UDP 213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa655559cda50ac3600b Record-Route: <sip:0123456789@212.112.162.50;ftag=2EBE3E60-1646;lr> Via: SIP/2.0/UDP 212.112.162.50;branch=z9hG4bK4885.ddcc862.0 Via: SIP/2.0/UDP 212.112.162.22:5060 From: <sip:9999999999@212.112.162.22>;tag=2EBE3E60-1646 To: <sip:0123456789@212.112.162.50> Date: Wed, 15 Dec 2004 10:10:11 GMT Call-ID: 56F68A14-4DB811D9-95A69155-12491096@212.112.162.22 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 1458717796-1303908825-2510524757-306778262 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 9 Remote-Party-ID: <sip:9999999999@212.112.162.22>;party=calling;screen=yes;privacy=off Timestamp: 1103105411 Contact: <sip:9999999999@212.112.162.22:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 288 v=0 o=CiscoSystemsSIP-GW-UserAgent 1486 6130 IN IP4 212.112.162.22 s=SIP Call c=IN IP4 212.112.162.22 t=0 0 m=audio 16842 RTP/AVP 18 0 101 c=IN IP4 212.112.162.22 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 24 headers, 12 lines Using latest request as basis request Sending to 213.132.103.213 : 5060 (non-NAT) Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 212.112.162.22:16842 Found description format G729 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Found peer 'wx3.se' Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa655559cda50ac3600b Via: SIP/2.0/UDP 212.112.162.50;branch=z9hG4bK4885.ddcc862.0 Via: SIP/2.0/UDP 212.112.162.22:5060 From: <sip:9999999999@212.112.162.22>;tag=2EBE3E60-1646 To: <sip:0123456789@212.112.162.50>;tag=as3c0db481 Call-ID: 56F68A14-4DB811D9-95A69155-12491096@212.112.162.22 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:s@192.1.1.1> Proxy-Authenticate: Digest realm="asterisk", nonce="59e60c89" Content-Length: 0 to 213.132.103.213:5060 Scheduling destruction of call '56F68A14-4DB811D9-95A69155-12491096@212.112.162.22' in 15000 ms Sip read: ACK sip:s@192.1.1.1 SIP/2.0 User-Agent: sapphire/1.6.2.0253 Max-Forwards: 70 Via: SIP/2.0/UDP 213.132.103.213:5060;branch=z9hG4bK-f73cba0b3f80aa655559cda50ac3600b To: <sip:0123456789@212.112.162.50>;tag=as3c0db481 From: <sip:9999999999@212.112.162.22>;tag=2EBE3E60-1646 Call-ID: 56F68A14-4DB811D9-95A69155-12491096@212.112.162.22 CSeq: 101 ACK Content-Length: 0 9 headers, 0 lines -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041216/2ea77417/attachment.htm