ошибки при вызове asterisk
поставил чистый астериск без веб примочек.
настраиваю с нуля.
| Код: |
| [general] register => nari904:pas@sip.rynga.com:5060/nari904 context=phones defaultexpiry=30 allowoverlap=no bindport=5060 srvlookup=yes allow=g729 allow=ulaw allow=alaw [rynga] type=peer host=sip.rynga.com fromuser=nari904 secret= context=phones dtmfmode=rfc2833 allow=g729 allow=ulaw allow=alaw insecure=invite [1000] type=friend context=phones host=dynamic fromuser=1000 secret=1234 callerid= |
extensions.conf
| Код: |
| [globals] OUTBOUNDTRUNK=sip/rynga [general] autofallthrough=yes [default] exten => s,1,Verbose(1|Unrouted call handler) exten => s,n,Answer() exten => s,n,Wait(1) exten => s,n,Playback(tt-weasels) exten => s,n,Hangup() [incoming_calls] [outgoing_calls] exten => 1000,1,Dial(${1000}) [internal] [outbound-local] exten => _NXXXXXXXXXX,1,Dial(sip/${EXTEN}@rynga) exten => _NXXXXXXXXXX,n,Congestion() exten => _NXXXXXXXXXX,n,Hangup() [phones] include => internal include => outbound-local exten => 1000,1,Dial(${1000}) |
Проблемы: 1. сделал в общей сложности в разные периоды времени 4 звонка. причем делаются звонки после каких-либо изменений в конфиге. звонок проходит, но слышимость односторонняя. после кладется трубка, и больше нельзя позвонить:
| Код: |
| [Oct 9 09:33:16] NOTICE[3925]: chan_sip.c:12063 sip_reregister: -- Re-registration for nari904@sip.rynga.com REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 77.72.174.128:5060: REGISTER sip:sip.rynga.com SIP/2.0 Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6deead74;rport Max-Forwards: 70 From: ;tag=as304dff58 To: Call-ID: 64b3bcd40fb5161970f909d1078334f0@172.24.26.8 CSeq: 215 REGISTER User-Agent: Asterisk PBX 1.6.2.22 Authorization: Digest username="nari904", realm="sip.rynga.com", algorithm=MD5, uri="sip:sip.rynga.com", nonce="636337953", response="e721639af66554f1d57c5e5fe453dbe5" Expires: 30 Contact: Content-Length: 0 --- SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6deead74;rport From: ;tag=as304dff58 To: Contact: ;expires=60 Call-ID: 64b3bcd40fb5161970f909d1078334f0@172.24.26.8 CSeq: 215 REGISTER Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 --- (10 headers 0 lines) --- Retransmitting #1 (NAT) to 77.72.174.128:5060: REGISTER sip:sip.rynga.com SIP/2.0 Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6deead74;rport Max-Forwards: 70 From: ;tag=as304dff58 To: Call-ID: 64b3bcd40fb5161970f909d1078334f0@172.24.26.8 CSeq: 215 REGISTER User-Agent: Asterisk PBX 1.6.2.22 Authorization: Digest username="nari904", realm="sip.rynga.com", algorithm=MD5, uri="sip:sip.rynga.com", nonce="636337953", response="e721639af66554f1d57c5e5fe453dbe5" Expires: 30 Contact: Content-Length: 0 --- SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6deead74;rport From: ;tag=as304dff58 To: Contact: ;expires=60 Call-ID: 64b3bcd40fb5161970f909d1078334f0@172.24.26.8 CSeq: 215 REGISTER Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 --- (10 headers 0 lines) --- Retransmitting #2 (NAT) to 77.72.174.128:5060: REGISTER sip:sip.rynga.com SIP/2.0 Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6deead74;rport Max-Forwards: 70 From: ;tag=as304dff58 To: Call-ID: 64b3bcd40fb5161970f909d1078334f0@172.24.26.8 CSeq: 215 REGISTER User-Agent: Asterisk PBX 1.6.2.22 Authorization: Digest username="nari904", realm="sip.rynga.com", algorithm=MD5, uri="sip:sip.rynga.com", nonce="636337953", response="e721639af66554f1d57c5e5fe453dbe5" Expires: 30 Contact: Content-Length: 0 --- SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6deead74;rport From: ;tag=as304dff58 To: Contact: ;expires=60 Call-ID: 64b3bcd40fb5161970f909d1078334f0@172.24.26.8 CSeq: 215 REGISTER Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 --- (10 headers 0 lines) --- Retransmitting #3 (NAT) to 77.72.174.128:5060: REGISTER sip:sip.rynga.com SIP/2.0 Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6deead74;rport Max-Forwards: 70 From: ;tag=as304dff58 To: Call-ID: 64b3bcd40fb5161970f909d1078334f0@172.24.26.8 CSeq: 215 REGISTER User-Agent: Asterisk PBX 1.6.2.22 Authorization: Digest username="nari904", realm="sip.rynga.com", algorithm=MD5, uri="sip:sip.rynga.com", nonce="636337953", response="e721639af66554f1d57c5e5fe453dbe5" Expires: 30 Contact: Content-Length: 0 --- SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6deead74;rport From: ;tag=as304dff58 To: Contact: ;expires=60 Call-ID: 64b3bcd40fb5161970f909d1078334f0@172.24.26.8 CSeq: 215 REGISTER Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 --- (10 headers 0 lines) --- Really destroying SIP dialog '64b3bcd40fb5161970f909d1078334f0@172.24.26.8' Method: REGISTER Retransmitting #4 (NAT) to 77.72.174.128:5060: REGISTER sip:sip.rynga.com SIP/2.0 Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6deead74;rport Max-Forwards: 70 From: ;tag=as304dff58 To: Call-ID: 64b3bcd40fb5161970f909d1078334f0@172.24.26.8 CSeq: 215 REGISTER User-Agent: Asterisk PBX 1.6.2.22 Authorization: Digest username="nari904", realm="sip.rynga.com", algorithm=MD5, uri="sip:sip.rynga.com", nonce="636337953", response="e721639af66554f1d57c5e5fe453dbe5" Expires: 30 Contact: Content-Length: 0 --- SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6deead74;rport From: ;tag=as304dff58 To: Contact: ;expires=60 Call-ID: 64b3bcd40fb5161970f909d1078334f0@172.24.26.8 CSeq: 215 REGISTER Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '64b3bcd40fb5161970f909d1078334f0@172.24.26.8' in 32000 ms (Method: REGISTER) [Oct 9 09:33:24] NOTICE[3925]: chan_sip.c:18875 handle_response_register: Outbound Registration: Expiry for sip.rynga.com is 30 sec (Scheduling reregistration in 23 s) Audio is at 172.24.26.8 port 18780 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 77.72.174.128:5060: INVITE sip:89046528473@sip.rynga.com SIP/2.0 Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6502d649;rport Max-Forwards: 70 From: "1000" ;tag=as1cd8af52 To: Contact: Call-ID: 1909fde735b9cd7b3b6a165f085150d3@172.24.26.8 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.22 Date: Tue, 09 Oct 2012 13:33:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 278 v=0 o=root 81935950 81935950 IN IP4 172.24.26.8 s=Asterisk PBX 1.6.2.22 c=IN IP4 172.24.26.8 t=0 0 m=audio 18780 RTP/AVP 0 8 3 101 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=ptime:20 a=sendrecv --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6502d649;rport From: "1000" ;tag=as1cd8af52 To: Contact: sip:89046528473@77.72.174.128:5060 Call-ID: 1909fde735b9cd7b3b6a165f085150d3@172.24.26.8 CSeq: 102 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE WWW-Authenticate: Digest realm="sipdiscount.com",nonce="636376484",algorithm=MD5 Content-Length: 0 --- (11 headers 0 lines) --- Transmitting (NAT) to 77.72.174.128:5060: ACK sip:89046528473@sip.rynga.com SIP/2.0 Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6502d649;rport Max-Forwards: 70 From: "1000" ;tag=as1cd8af52 To: Contact: Call-ID: 1909fde735b9cd7b3b6a165f085150d3@172.24.26.8 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.22 Content-Length: 0 --- [Oct 9 09:33:31] NOTICE[3925]: chan_sip.c:18458 handle_response_invite: Failed to authenticate on INVITE to '"1000" ;tag=as1cd8af52' Really destroying SIP dialog '1909fde735b9cd7b3b6a165f085150d3@172.24.26.8' Method: INVITE |
Подскажите пожалуйста, почему не проходит регистрация? или слетает она?
пользуйтесь поиском наконец
есть у кого то соображения по этому поводу?+
добавление строк externip loacalnet nat не принесло результата