(this time with subject....) Hello,
Im trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because Im new to Asterisk I cant get the error why this is not working. To me it all looks fine, no warnings or what so ever The settings in sip.conf and extensions.conf are identical to those of http://www.voip-info.org/tiki-index.php?page=Talkin2ya Does anyone know what Im doing wrong???? Thanks, Peter. ------------------------------- output of sip debug ------------------------------- 11 headers, 0 lines Reliably Transmitting (no NAT) to 81.23.228.150:5060: REGISTER sip:budgetphone.nl SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc From: <sip:[EMAIL PROTECTED]>;tag=as5dc83db4 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: <sip:[EMAIL PROTECTED]> Event: registration Content-Length: 0 --- server*CLI> <-- SIP read from 81.23.228.150:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc From: <sip:[EMAIL PROTECTED]>;tag=as5dc83db4 To: <sip:[EMAIL PROTECTED]>;tag=9b5971f23d18872ff678d4e9dae023f8.247a Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER WWW-Authenticate: Digest realm="budgetphone.nl", nonce="42d15009299d7652e8da589cee2723af4b6a96ca" Server: Sip EXpress router (0.8.14-5 (i386/linux)) Content-Length: 0 --- (9 headers 0 lines)--- Responding to challenge, registration to domain/host name budgetphone.nl 12 headers, 0 lines Reliably Transmitting (no NAT) to 81.23.228.150:5060: REGISTER sip:budgetphone.nl SIP/2.0 Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e From: <sip:[EMAIL PROTECTED]>;tag=as7e56000d To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="31717110342", realm="budgetphone.nl", algorithm=MD5, uri="sip:budgetphone.nl", nonce="42d15009299d7652e8da589cee2723af4b6a96ca", response="cd69279e6a2512fd48d267ceea3394da", opaque="" Expires: 120 Contact: <sip:[EMAIL PROTECTED]> Event: registration Content-Length: 0 --- server*CLI> <-- SIP read from 81.23.228.150:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e From: <sip:[EMAIL PROTECTED]>;tag=as7e56000d To: <sip:[EMAIL PROTECTED]>;tag=9b5971f23d18872ff678d4e9dae023f8.98b0 Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER Contact: <sip:[EMAIL PROTECTED]:5060>;q=0.00;expires=120 Server: Sip EXpress router (0.8.14-5 (i386/linux)) Content-Length: 0 --- (9 headers 0 lines)--- Jul 10 18:38:04 NOTICE[26004]: chan_sip.c:8266 handle_response: Outbound Registration: Expiry for budgetphone.nl is 120 sec (Scheduling reregistration in 105000 ms) Destroying call '[EMAIL PROTECTED]' server*CLI> <-- SIP read from 81.23.228.150:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Max-Forwards: 10 Record-Route: <sip:[EMAIL PROTECTED];ftag=as47419911;lr=on> Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0 Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa From: "0031172651375" <sip:[EMAIL PROTECTED]>;tag=as47419911 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sun, 10 Jul 2005 16:37:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 345 v=0 o=root 26318 26318 IN IP4 212.203.28.2 s=session c=IN IP4 81.23.228.139 t=0 0 m=audio 36634 RTP/AVP 3 18 5 0 97 110 101 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 15 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 81.23.228.150 : 5060 (NAT) Found peer '31717110342' Reliably Transmitting (NAT) to 81.23.228.150:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0;received=81.23.228.150;rport=506 0 Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa From: "0031172651375" <sip:[EMAIL PROTECTED]>;tag=as47419911 To: <sip:[EMAIL PROTECTED]>;tag=as3f35655f Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Proxy-Authenticate: Digest realm="asterisk", nonce="555b996d" Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms server*CLI> <-- SIP read from 81.23.228.150:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0 From: "0031172651375" <sip:[EMAIL PROTECTED]>;tag=as47419911 Call-ID: [EMAIL PROTECTED] To: <sip:[EMAIL PROTECTED]>;tag=as3f35655f CSeq: 102 ACK User-Agent: Sip EXpress router(0.8.14-5 (i386/linux)) Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' server*CLI> _______________________________________________ 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
