ช่วยด้วยครับ regiser ระหว่าง voip gateway กับ asterisk ไม่ได
โพสต์แล้ว: 15 มี.ค. 2011 12:46
ผมลอง debug ดูได้ข้อมูลตามนี้ครับ
ลองสั่ง show registry จะได้ตามนี้ครับ
คำสั่ง show peers ได้ตามนี้ครับ
ในหน้า web config ของ voip
set sip server address กับ Outbound Proxy Address เป็น 192.168.1.130 port 5060
192.168.1.24 เป็น voip gateway
192.168.1.130 เป็นตัว asterisk
- โค้ด: เลือกทั้งหมด
<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '4cdad419244945d657440d7748bb0288@192.168.1.130' Method: OPTIONS
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.24:5060:
REGISTER sip:192.168.1.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK741fee89;rport
Max-Forwards: 70
From: <sip:6000@192.168.1.24>;tag=as090a2f15
To: <sip:6000@192.168.1.24>
Call-ID: 3724f5a23a0a50812622b72f7860e20c@127.0.0.1
CSeq: 153 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Expires: 120
Contact: <sip:s@192.168.1.130>
Content-Length: 0
---
Really destroying SIP dialog '3724f5a23a0a50812622b72f7860e20c@127.0.0.1' Method: REGISTER
Retransmitting #1 (no NAT) to 192.168.1.24:5060:
REGISTER sip:192.168.1.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK741fee89;rport
Max-Forwards: 70
From: <sip:6000@192.168.1.24>;tag=as090a2f15
To: <sip:6000@192.168.1.24>
Call-ID: 3724f5a23a0a50812622b72f7860e20c@127.0.0.1
CSeq: 153 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Expires: 120
Contact: <sip:s@192.168.1.130>
Content-Length: 0
---
Retransmitting #2 (no NAT) to 192.168.1.24:5060:
REGISTER sip:192.168.1.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK741fee89;rport
Max-Forwards: 70
From: <sip:6000@192.168.1.24>;tag=as090a2f15
To: <sip:6000@192.168.1.24>
Call-ID: 3724f5a23a0a50812622b72f7860e20c@127.0.0.1
CSeq: 153 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Expires: 120
Contact: <sip:s@192.168.1.130>
Content-Length: 0
---
Retransmitting #3 (no NAT) to 192.168.1.24:5060:
REGISTER sip:192.168.1.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK741fee89;rport
Max-Forwards: 70
From: <sip:6000@192.168.1.24>;tag=as090a2f15
To: <sip:6000@192.168.1.24>
Call-ID: 3724f5a23a0a50812622b72f7860e20c@127.0.0.1
CSeq: 153 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Expires: 120
Contact: <sip:s@192.168.1.130>
Content-Length: 0
---
<--- SIP read from UDP:192.168.1.24:5060 --->
<------------->
Retransmitting #4 (no NAT) to 192.168.1.24:5060:
REGISTER sip:192.168.1.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK741fee89;rport
Max-Forwards: 70
From: <sip:6000@192.168.1.24>;tag=as090a2f15
To: <sip:6000@192.168.1.24>
Call-ID: 3724f5a23a0a50812622b72f7860e20c@127.0.0.1
CSeq: 153 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Expires: 120
Contact: <sip:s@192.168.1.130>
Content-Length: 0
---
Retransmitting #5 (no NAT) to 192.168.1.24:5060:
REGISTER sip:192.168.1.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK741fee89;rport
Max-Forwards: 70
From: <sip:6000@192.168.1.24>;tag=as090a2f15
To: <sip:6000@192.168.1.24>
Call-ID: 3724f5a23a0a50812622b72f7860e20c@127.0.0.1
CSeq: 153 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Expires: 120
Contact: <sip:s@192.168.1.130>
Content-Length: 0
---
Retransmitting #6 (no NAT) to 192.168.1.24:5060:
REGISTER sip:192.168.1.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:5060;branch=z9hG4bK741fee89;rport
Max-Forwards: 70
From: <sip:6000@192.168.1.24>;tag=as090a2f15
To: <sip:6000@192.168.1.24>
Call-ID: 3724f5a23a0a50812622b72f7860e20c@127.0.0.1
CSeq: 153 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Expires: 120
Contact: <sip:s@192.168.1.130>
Content-Length: 0
---
ลองสั่ง show registry จะได้ตามนี้ครับ
- โค้ด: เลือกทั้งหมด
elastix*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
sip.kallkool.com:5060 N 0027600839 105 Registered Tue, 15 Mar 2011 12:22:35
sip.skype.com:5060 N 990510001314 105 Registered Tue, 15 Mar 2011 12:22:27
192.168.1.24:5060 N 6000 120 Request Sent
3 SIP registrations.
คำสั่ง show peers ได้ตามนี้ครับ
- โค้ด: เลือกทั้งหมด
elastix*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
6000/6000 192.168.1.24 D N A 5060 OK (22 ms)
Skype/99051000131441 204.9.161.164 5060 Unmonitored
ThaiTelephone/0027600839 61.90.185.178 5060 OK (94 ms)
gsmtest/6000 192.168.1.24 N 5060 OK (127 ms)
4 sip peers [Monitored: 3 online, 0 offline Unmonitored: 1 online, 0 offline]
ในหน้า web config ของ voip
set sip server address กับ Outbound Proxy Address เป็น 192.168.1.130 port 5060
192.168.1.24 เป็น voip gateway
192.168.1.130 เป็นตัว asterisk