Did you port forwar in your router RTP ports ? 10000-20000 to your *Box ?
On 7/21/06, Jose Limeres <[EMAIL PROTECTED]> wrote:
Hi, I am experiencing a hard to solve problem with my VoIP provider. I can make calls without any problem but I can not receive any. Actually, calls arive to * but the phone just does not ring. I believe must be a problem with NAT but I think I have a good config: - Extensions have nat=always and qualify=yes - Have introduced in sip.conf Externip and localnet - ADSL modem/router is redirected to my * server - With sip debug I can see the call arrives Am I misssing something that someone else can see? Appreciate any hint. Thanks ============================== ====== ASTERISK VERSION: Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q SIP DEBUG CAPTURE <-- SIP read from 62.22.20.194:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Record-Route: <sip: 62.22.20.194;ftag=08ff6000ff05ff10ff00000e0c4effff;lr> Via: SIP/2.0/UDP 62.22.20.194;branch=z9hG4bK90bf.b9c560e1.0 Via: SIP/2.0/UDP 62.22.20.207:5060;branch=z9hG4bK-10ff6000ff05ff10ff00000e0c4effff From: <sip:[EMAIL PROTECTED];user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff To: < sip:[EMAIL PROTECTED]:5060;user=phone> Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: < sip:[EMAIL PROTECTED];user=phone> Max-Forwards: 9 User-Agent: MERA MSIP v.1.0.2 Cisco-Guid: 908093991-393679323-3151091529-1429652222 Content-Type: application/sdp Content-Length: 216 v=0 o=- 1153435071 1153435071 IN IP4 62.22.20.207 s=- c=IN IP4 62.22.20.207 t=0 0 m=audio 59320 RTP/AVP 18 4 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (14 headers 10 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 62.22.20.194 : 5060 (non-NAT) Found peer 'Peoplecall' Reliably Transmitting (NAT) to 62.22.20.194:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 62.22.20.194;branch=z9hG4bK90bf.b9c560e1.0;received= 62.22.20.194 Via: SIP/2.0/UDP 62.22.20.207:5060;branch=z9hG4bK-10ff6000ff05ff10ff00000e0c4effff From: < sip:[EMAIL PROTECTED];user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff To: <sip:[EMAIL PROTECTED] :5060;user=phone>;tag=as476d14de Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: < sip:[EMAIL PROTECTED]> Proxy-Authenticate: Digest realm="asterisk", nonce="008d23b0" Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms asterisk1*CLI> <-- SIP read from 62.22.20.194:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 62.22.20.194;branch= z9hG4bK90bf.b9c560e1.0 From: <sip:[EMAIL PROTECTED];user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff Call-ID: [EMAIL PROTECTED] To: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=as476d14de CSeq: 1 ACK User-Agent: OpenSer (1.0.0 (i386/linux)) Content-Length: 0 --- (8 headers 0 lines)--- REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 62.22.20.194:5060 : REGISTER sip:sip.peoplecall.com SIP/2.0 Via: SIP/2.0/UDP 87.218.175.74:5060;branch=z9hG4bK4a6abe4f;rport From: <sip:[EMAIL PROTECTED] >;tag=as79fdfc26 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 421 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="34700758288001", realm=" sip.peoplecall.com", algorithm=MD5, uri="sip:sip.peoplecall.com ", nonce="44c0059db2d71f523aeb30399a54a4a32d8aeed6", response="ee782a37bae7eed1a0a881147c733ede", opaque="" Expires: 120 Contact: <sip:[EMAIL PROTECTED]> Event: registration Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 62.22.20.194:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK4a6abe4f;rport=5060 From: < sip:[EMAIL PROTECTED]>;tag=as79fdfc26 To: <sip:[EMAIL PROTECTED] >;tag=555271b30cfd40f8a3b4837b054360a3.975d Call-ID: [EMAIL PROTECTED] CSeq: 421 REGISTER Contact: <sip:[EMAIL PROTECTED]:5060>;expires=120 Server: OpenSer (1.0.0 (i386/linux)) Content-Length: 0 --- (9 headers 0 lines)--- Scheduling destruction of call ' [EMAIL PROTECTED]' in 32000 ms Destroying call '[EMAIL PROTECTED] ' asterisk1*CLI> sip no debug SIP Debugging Disabled _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- Com os melhores cumprimentos, Marco Mouta _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
