asterisk.ru

Код: выделить все

provider/login1          ip                              N          A  5068     Unmonitored
provider2/login2         ip                              N          A  5068     Unmonitored
provider3/login3         ip                              N          A  5068     Unmonitored

ded писал(а):john, потому что для системы, где регистрируетесь - ip:port
вы представляетесь единственным ip:port, неважно, что разные логины.

Код: выделить все

<--- SIP read from TLS:172.22.12.140:12123 --->


<------------->
  == Using SIP RTP CoS mark 5
    -- Executing [number03@incoming1:1] Goto("SIP/provider-0000042e", "supercontext,number03,1") in new stack
    -- Goto (supercontext,number03,1)
    -- Executing [number03@supercontext:1] Dial("SIP/provider-0000042e", "SIP/8003,25,tm") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 14466
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.22.12.140:12123:
INVITE sip:8003@172.22.12.140:12123;transport=TLS SIP/2.0
Via: SIP/2.0/TLS ipServer:5061;branch=z9hG4bK488ac70f;rport
Max-Forwards: 70
From: "mobileNumber" <sip:mobileNumber@ipServer>;tag=as36026af0
To: <sip:8003@172.22.12.140:12123;transport=TLS>
Contact: <sip:mobileNumber@ipServer:5061;transport=TLS>
Call-ID: 107f5a9e374ddd8c5add439d05031363@ipServer:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.32.3
Date: Thu, 05 Sep 2019 10:20:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 884449564 884449564 IN IP4 ipServer
s=Asterisk PBX 1.8.32.3
c=IN IP4 ipServer
t=0 0
m=audio 14466 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/8003
    -- Started music on hold, class 'default', on channel 'SIP/provider-0000042e'

<--- SIP read from TLS:172.22.12.140:12123 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS ipServer:5061;branch=z9hG4bK488ac70f;rport=5061
From: "mobileNumber" <sip:mobileNumber@ipServer>;tag=as36026af0
To: <sip:8003@172.22.12.140:12123;transport=TLS>
Call-ID: 107f5a9e374ddd8c5add439d05031363@ipServer:5061
CSeq: 102 INVITE
User-Agent: Yealink SIP-T21P_E2 52.81.0.70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from TLS:172.22.12.140:12123 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/TLS ipServer:5061;branch=z9hG4bK488ac70f;rport=5061
From: "mobileNumber" <sip:mobileNumber@ipServer>;tag=as36026af0
To: <sip:8003@172.22.12.140:12123;transport=TLS>;tag=1810026559
Call-ID: 107f5a9e374ddd8c5add439d05031363@ipServer:5061
CSeq: 102 INVITE
User-Agent: Yealink SIP-T21P_E2 52.81.0.70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 172.22.12.140:12123:
ACK sip:8003@172.22.12.140:12123;transport=TLS SIP/2.0
Via: SIP/2.0/TLS ipServer:5061;branch=z9hG4bK488ac70f;rport
Max-Forwards: 70
From: "mobileNumber" <sip:mobileNumber@ipServer>;tag=as36026af0
To: <sip:8003@172.22.12.140:12123;transport=TLS>;tag=1810026559
Contact: <sip:mobileNumber@ipServer:5061;transport=TLS>
Call-ID: 107f5a9e374ddd8c5add439d05031363@ipServer:5061
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.32.3
Content-Length: 0


---
Scheduling destruction of SIP dialog '107f5a9e374ddd8c5add439d05031363@ipServer:5061' in 32000 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Stopped music on hold on SIP/provider-0000042e
    -- Auto fallthrough, channel 'SIP/provider-0000042e' status is 'CHANUNAVAIL'

Код: выделить все

exten => number03,1,Goto(supercontext,${EXTEN},1)

[supercontext]

exten => number03,1,Set(CHANNEL(secure_bridge_signaling)=1)
same => n,Set(CHANNEL(secure_bridge_media)=1)
same => n,Dial(SIP/8003,25,tm)
same => n,Set(CHANNEL(secure_bridge_signaling)=0)
same => n,Set(CHANNEL(secure_bridge_media)=0)
same => n,WaitExten(5)