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> 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 
