Hi everybody, I was not successful to make my Asteirsk receive calls. Please help me to set it up. My Asterisk is behind a Linksys router. My Local IP, i.e. Router's IP is 192.168.0.1 My External IP is 24.57.xxx.xxx My Asterisk's IP is 192.168.0.105 My X-Lite's IP is 192.168.0.103 and it is on ext 201 My SIP provider's IP is 209.167.xxx.xxx My extensions work fine, and X-Lite also successfully dials out. But the problem begins when I dial to my Asterisk server from my other phone which is 514-854-7804 On debugging, I get the following screen which shows that it is successfully receiving the call but is not directing it to my extension 201, which is X-Lite. =========================== Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 209.167.xxx.xxx:5060;branch=z9hG4bK6cfbf611;rport From: "5148547804" ;tag=as664e8fb1 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sun, 13 Nov 2005 06:12:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 377 v=0 o=root 2173 2173 IN IP4 209.167.xxx.xxx s=session c=IN IP4 209.167.xxx.xxx t=0 0 m=audio 11448 RTP/AVP 18 97 110 111 0 8 3 101 a=rtpmap:18 G729/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - Nov 12 22:03:00 VERBOSE[1430]: 12 headers, 16 lines Nov 12 22:03:00 DEBUG[1430]: ##### Testing 209.167.xxx.xxx with 192.168.0.0 Nov 12 22:03:00 DEBUG[1430]: Target address 209.167.xxx.xxx is not local, substituting externip Nov 12 22:03:00 VERBOSE[1430]: Using latest request as basis request Nov 12 22:03:00 VERBOSE[1430]: Sending to 209.167.xxx.xxx : 5060 (NAT) Nov 12 22:03:00 VERBOSE[1430]: Found peer '209.167.xxx.xxx' Nov 12 22:03:00 DEBUG[1430]: Setting NAT on RTP to 0 Nov 12 22:03:00 VERBOSE[1430]: Found RTP audio format 18 Nov 12 22:03:00 VERBOSE[1430]: Found RTP audio format 97 Nov 12 22:03:00 VERBOSE[1430]: Found RTP audio format 110 Nov 12 22:03:00 VERBOSE[1430]: Found RTP audio format 111 Nov 12 22:03:00 VERBOSE[1430]: Found RTP audio format 0 Nov 12 22:03:00 VERBOSE[1430]: Found RTP audio format 8 Nov 12 22:03:00 VERBOSE[1430]: Found RTP audio format 3 Nov 12 22:03:00 VERBOSE[1430]: Found RTP audio format 101 Nov 12 22:03:00 VERBOSE[1430]: Peer audio RTP is at port 209.167.xxx.xxx:11448 Nov 12 22:03:00 DEBUG[1430]: Peer audio RTP is at port 209.167.xxx.xxx:11448 Nov 12 22:03:00 VERBOSE[1430]: Found description format G729 Nov 12 22:03:00 VERBOSE[1430]: Found description format iLBC Nov 12 22:03:00 VERBOSE[1430]: Found description format speex Nov 12 22:03:00 VERBOSE[1430]: Found description format G726-32 Nov 12 22:03:00 VERBOSE[1430]: Found description format PCMU Nov 12 22:03:00 VERBOSE[1430]: Found description format PCMA Nov 12 22:03:00 VERBOSE[1430]: Found description format GSM Nov 12 22:03:00 VERBOSE[1430]: Found description format telephone-event Nov 12 22:03:00 VERBOSE[1430]: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x71e (gsm|ulaw|alaw|g726|g729|speex|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Nov 12 22:03:00 VERBOSE[1430]: Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Nov 12 22:03:00 DEBUG[1430]: Check for res for Nov 12 22:03:00 DEBUG[1430]: is not a local user Nov 12 22:03:00 VERBOSE[1430]: Looking for s in ravi Nov 12 22:03:00 VERBOSE[1430]: Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 209.167.xxx.xxx:5060;branch=z9hG4bK6cfbf611 From: "5148547804" ;tag=as664e8fb1 To: ;tag=as10f3cdea Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 =========================== My sip.conf is =========================== [general] port = 5060 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw nat=yes externip=24.57.xxx.xxx fromdomain=209.167.xxx.xxx localnet=192.168.0.1/255.255.255.0 context = external register=555555558:[EMAIL PROTECTED] [201] username=201 type=friend secret=1111 qualify=no port=5060 nat=yes host=dynamic dtmfmode=rfc2833 context=internal canreinvite=no callerid="Home Phone" <201> [trunk] username=555555558 type=peer secret=47076 host=209.167.xxx.xxx [trunk_incoming] type=user secret=47076 host=209.167.xxx.xxx =========================== My extensions.conf has following lines to deal with incoming SIP calls =========================== [external] exten => _X.,1,Dial(SIP/201,60,tr)
=========================== Please tell me where am i making mistake. If somebody else has same setup as mines, i'll appreciate if you can send me your sip.conf and extensions.conf Thanks, Zeeshan A Zakaria _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- 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
