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ошибки при исходящем вызове asterisk

https://forum.asterisk.ru/viewtopic.php?t=2129

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ошибки при исходящем вызове asterisk

Добавлено: 09 окт 2012, 15:59
rayden8
Доброго дня, коллеги.
поставил чистый астериск без веб примочек.
настраиваю с нуля. До этого пользовался только веб мордами

sip.conf

Код: Выделить всё

[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=<pas> 
context=phones 
dtmfmode=rfc2833 
allow=g729 
allow=ulaw 
allow=alaw 
insecure=invite 


[1000] 
type=friend 
context=phones 
host=dynamic 
fromuser=1000 
secret=1234 
callerid=<1000>
extensions.conf

Код: Выделить всё

[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: Выделить всё

[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: <sip:nari904@sip.rynga.com>;tag=as304dff58 
To: <sip:nari904@sip.rynga.com> 
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: <sip:nari904@172.24.26.8> 
Content-Length: 0 


--- 

<--- SIP read from UDP:77.72.174.128:5060 ---> 
SIP/2.0 200 Ok 
Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6deead74;rport 
From: <sip:nari904@sip.rynga.com>;tag=as304dff58 
To: <sip:nari904@sip.rynga.com> 
Contact: <sip:nari904@172.24.26.8:5060>;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: <sip:nari904@sip.rynga.com>;tag=as304dff58 
To: <sip:nari904@sip.rynga.com> 
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: <sip:nari904@172.24.26.8> 
Content-Length: 0 


--- 

<--- SIP read from UDP:77.72.174.128:5060 ---> 
SIP/2.0 200 Ok 
Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6deead74;rport 
From: <sip:nari904@sip.rynga.com>;tag=as304dff58 
To: <sip:nari904@sip.rynga.com> 
Contact: <sip:nari904@172.24.26.8:5060>;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: <sip:nari904@sip.rynga.com>;tag=as304dff58 
To: <sip:nari904@sip.rynga.com> 
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: <sip:nari904@172.24.26.8> 
Content-Length: 0 


--- 

<--- SIP read from UDP:77.72.174.128:5060 ---> 
SIP/2.0 200 Ok 
Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6deead74;rport 
From: <sip:nari904@sip.rynga.com>;tag=as304dff58 
To: <sip:nari904@sip.rynga.com> 
Contact: <sip:nari904@172.24.26.8:5060>;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: <sip:nari904@sip.rynga.com>;tag=as304dff58 
To: <sip:nari904@sip.rynga.com> 
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: <sip:nari904@172.24.26.8> 
Content-Length: 0 


--- 

<--- SIP read from UDP:77.72.174.128:5060 ---> 
SIP/2.0 200 Ok 
Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6deead74;rport 
From: <sip:nari904@sip.rynga.com>;tag=as304dff58 
To: <sip:nari904@sip.rynga.com> 
Contact: <sip:nari904@172.24.26.8:5060>;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: <sip:nari904@sip.rynga.com>;tag=as304dff58 
To: <sip:nari904@sip.rynga.com> 
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: <sip:nari904@172.24.26.8> 
Content-Length: 0 


--- 

<--- SIP read from UDP:77.72.174.128:5060 ---> 
SIP/2.0 200 Ok 
Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6deead74;rport 
From: <sip:nari904@sip.rynga.com>;tag=as304dff58 
To: <sip:nari904@sip.rynga.com> 
Contact: <sip:nari904@172.24.26.8:5060>;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" <sip:nari904@172.24.26.8>;tag=as1cd8af52 
To: <sip:89046528473@sip.rynga.com> 
Contact: <sip:nari904@172.24.26.8> 
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 read from UDP:77.72.174.128:5060 ---> 
SIP/2.0 401 Unauthorized 
Via: SIP/2.0/UDP 172.24.26.8:5060;branch=z9hG4bK6502d649;rport 
From: "1000" <sip:nari904@172.24.26.8:5060>;tag=as1cd8af52 
To: <sip:89046528473@sip.rynga.com> 
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" <sip:nari904@172.24.26.8>;tag=as1cd8af52 
To: <sip:89046528473@sip.rynga.com> 
Contact: <sip:nari904@172.24.26.8> 
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" <sip:nari904@172.24.26.8>;tag=as1cd8af52' 
Really destroying SIP dialog '1909fde735b9cd7b3b6a165f085150d3@172.24.26.8' Method: INVITE
Подскажите пожалуйста, почему не проходит регистрация? или слетает она?

Re: ошибки при исходящем вызове asterisk

Добавлено: 09 окт 2012, 17:17
ded
Изображение

Re: ошибки при исходящем вызове asterisk

Добавлено: 11 окт 2012, 12:13
rayden8
Спасибо
Часовой пояс: UTC+03:00
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