14-06-07, Marcin Kawelski <[EMAIL PROTECTED]> napisał(a):
. Jeśli próbowałeś i nic to opisz dokładniej problem...
Dokładnie to sprawa wygląda tak : musi toto stać na [EMAIL PROTECTED] wszystkie połączenia muszą być skonfigurowane przez freePBX a nie wpisane na sztywno do sip.conf Połączenia wychodzące i między pozostałymi telefonami działają bez zarzutu Asterisk stoi na publicznym IP, więc nie ma problemów z natowaniem (bo to podejrzewałem na początku) Całkiem możliwe że po prostu źle coś wpisuję do panelu freePBX Poniżej fragment z debugu: --- (0 headers 1 lines)--- asterisk1*CLI> <-- SIP read from 195.200.214.111:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 195.200.214.111:5060;branch=z9hG4bK12688009 From: <sip:[EMAIL PROTECTED]>;tag=2a8911b4 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Contact: <sip:[EMAIL PROTECTED]> Max-Forwards: 70 Allow: INVITE, ACK, BYE, CANCEL, REGISTER Content-Type: application/sdp Content-Length: 366 v=0 o=- 18976717 18976717 IN IP4 83.14.53.50 s=- c=IN IP4 195.200.214.111 t=0 0 m=audio 15682 RTP/AVP 18 0 2 8 98 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (11 headers 17 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 195.200.214.111 : 5060 (non-NAT) Found peer 'tlenwawa' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 8 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 195.200.214.111:15682 Found description format G729a Found description format PCMU Found description format G726-32 Found description format PCMA Found description format G726-16 Found description format NSE Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x11c (ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for ls-fli00fb27224-2 in from-tlenofon (domain sip.tlenofon.pl) Reliably Transmitting (no NAT) to 195.200.214.111:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 195.200.214.111:5060;branch=z9hG4bK12688009;received= 195.200.214.111 From: <sip:[EMAIL PROTECTED]>;tag=2a8911b4 To: <sip:[EMAIL PROTECTED]>;tag=as6dbc55d1 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 195.200.214.111:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 195.200.214.111:5060;branch=z9hG4bK12688009 From: <sip:[EMAIL PROTECTED]>;tag=2a8911b4 To: <sip:[EMAIL PROTECTED]>;tag=as6dbc55d1 Call-ID: [EMAIL PROTECTED] CSeq: 101 ACK Content-Length: 0
