|
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:[EMAIL PROTECTED]
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). Sip.conf: [general] context=demo [wx3.se] 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:[EMAIL PROTECTED] 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:[EMAIL PROTECTED];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:[EMAIL PROTECTED]>;tag=2EBE3E60-1646 To: <sip:[EMAIL PROTECTED]> Date: Wed, 15 Dec 2004 10:10:11 GMT Call-ID:
[EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;party=calling;screen=yes;privacy=off Timestamp: 1103105411 Contact: <sip:[EMAIL PROTECTED]:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 288 v=0 o=CiscoSystemsSIP-GW-UserAgent 1486 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:[EMAIL PROTECTED]>;tag=2EBE3E60-1646 To: <sip:[EMAIL PROTECTED]>;tag=as3c0db481 Call-ID:
[EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Proxy-Authenticate: Digest
realm="asterisk", nonce="59e60c89" Content-Length: 0 �to 213.132.103.213:5060 Scheduling destruction of call
'[EMAIL PROTECTED]' in 15000 ms Sip read: ACK sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=as3c0db481 From: <sip:[EMAIL PROTECTED]>;tag=2EBE3E60-1646 Call-ID:
[EMAIL PROTECTED] CSeq: 101 ACK Content-Length: 0 9 headers, 0 lines |
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
