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Got SIP response 489 "Bad event" back from แก้ยังไงครับ

Asterisk Opensource IP Pbx
ตอบกระทู้

Re: Got SIP response 489 "Bad event" back from แก้ยังไงครับ

โพสต์โดย apc » 30 ก.ย. 2011 07:54

มันยาวหน่อยนะครับ

โค้ด: เลือกทั้งหมด
Max-Forwards: 70
From: <sip:22345@192.168.1.112>;tag=as1c38d18a
To: <sip:22345@192.168.1.112>
Call-ID: 092bda326da00ba56e5ee8407782741f@192.168.1.112
CSeq: 123 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Authorization: Digest username="22345", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.112", nonce="57d35988", response="9ec71bb9918215a7a9e2694bfb0fcef7"
Expires: 120
Contact: <sip:22345@192.168.1.111>
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.111 : 5060 (NAT)

<--- Transmitting (NAT) to 192.168.1.111:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK08a29aea;received=192.168.1.111;rport=5060
From: <sip:22345@192.168.1.112>;tag=as1c38d18a
To: <sip:22345@192.168.1.112>
Call-ID: 092bda326da00ba56e5ee8407782741f@192.168.1.112
CSeq: 123 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:22345@192.168.1.112>
Content-Length: 0


<------------>

<--- Transmitting (NAT) to 192.168.1.111:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK08a29aea;received=192.168.1.111;rport=5060
From: <sip:22345@192.168.1.112>;tag=as1c38d18a
To: <sip:22345@192.168.1.112>;tag=as52eec899
Call-ID: 092bda326da00ba56e5ee8407782741f@192.168.1.112
CSeq: 123 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 120
Contact: <sip:22345@192.168.1.111>;expires=120
Date: Fri, 30 Sep 2011 00:51:08 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '092bda326da00ba56e5ee8407782741f@192.168.1.112' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '05278f9a318b5ba35cba0f6f3892162b@192.168.1.111' Method: OPTIONS
Really destroying SIP dialog '477e529d3914d7cc2680e64f6675eefd@192.168.1.111' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.1.111:5060:
OPTIONS sip:22345@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK4fb5d184;rport
From: "Unknown" <sip:Unknown@192.168.1.112>;tag=as69bef4e3
To: <sip:22345@192.168.1.111>
Contact: <sip:Unknown@192.168.1.112>
Call-ID: 0f24e85d6b583ddd7f1209ca32f72cfa@192.168.1.112
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 30 Sep 2011 00:51:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Scheduling destruction of SIP dialog '56854c897b424cde144b78c56b40daa2@192.168.1.112' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.1.111:5060:
NOTIFY sip:22345@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK64820e8b;rport
From: "Unknown" <sip:Unknown@192.168.1.112>;tag=as083fe03d
To: <sip:22345@192.168.1.111>
Contact: <sip:Unknown@192.168.1.112>
Call-ID: 56854c897b424cde144b78c56b40daa2@192.168.1.112
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*97@192.168.1.112
Voice-Message: 0/0 (0/0)

---
trixbox1*CLI>
<--- SIP read from 192.168.1.111:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK4fb5d184;received=192.168.1.112;rport=5060
From: "Unknown" <sip:Unknown@192.168.1.112>;tag=as69bef4e3
To: <sip:22345@192.168.1.111>;tag=as1e8ccba0
Call-ID: 0f24e85d6b583ddd7f1209ca32f72cfa@192.168.1.112
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.1.111>
Accept: application/sdp
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '0f24e85d6b583ddd7f1209ca32f72cfa@192.168.1.112' Method: OPTIONS
trixbox1*CLI>
<--- SIP read from 192.168.1.111:5060 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK64820e8b;received=192.168.1.112;rport=5060
From: "Unknown" <sip:Unknown@192.168.1.112>;tag=as083fe03d
To: <sip:22345@192.168.1.111>;tag=as19764809
Call-ID: 56854c897b424cde144b78c56b40daa2@192.168.1.112
CSeq: 102 NOTIFY
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
    -- Got SIP response 489 "Bad event" back from 192.168.1.111
Really destroying SIP dialog '56854c897b424cde144b78c56b40daa2@192.168.1.112' Method: NOTIFY
trixbox1*CLI>
<--- SIP read from 192.168.1.111:5060 --->
OPTIONS sip:192.168.1.112 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK16888007;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.111>;tag=as08840912
To: <sip:192.168.1.112>
Contact: <sip:Unknown@192.168.1.111>
Call-ID: 5368afd0165f78f54bae898e1baef8c4@192.168.1.111
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Fri, 30 Sep 2011 00:51:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Looking for s in from-sip-external (domain 192.168.1.112)

<--- Transmitting (no NAT) to 192.168.1.111:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK16888007;received=192.168.1.111;rport=5060
From: "Unknown" <sip:Unknown@192.168.1.111>;tag=as08840912
To: <sip:192.168.1.112>;tag=as108b7a42
Call-ID: 5368afd0165f78f54bae898e1baef8c4@192.168.1.111
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:192.168.1.112>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '5368afd0165f78f54bae898e1baef8c4@192.168.1.111' in 32000 ms (Method: OPTIONS)
trixbox1*CLI>
<--- SIP read from 192.168.1.111:5060 --->
OPTIONS sip:192.168.1.112 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK4a6cad8f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.111>;tag=as6026a8fa
To: <sip:192.168.1.112>
Contact: <sip:Unknown@192.168.1.111>
Call-ID: 1953a3d55cae0cb05c933668227f9919@192.168.1.111
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Fri, 30 Sep 2011 00:51:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Looking for s in from-sip-external (domain 192.168.1.112)

<--- Transmitting (no NAT) to 192.168.1.111:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK4a6cad8f;received=192.168.1.111;rport=5060
From: "Unknown" <sip:Unknown@192.168.1.111>;tag=as6026a8fa
To: <sip:192.168.1.112>;tag=as0cf0506b
Call-ID: 1953a3d55cae0cb05c933668227f9919@192.168.1.111
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:192.168.1.112>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1953a3d55cae0cb05c933668227f9919@192.168.1.111' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '092bda326da00ba56e5ee8407782741f@192.168.1.112' Method: REGISTER
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.111:5060:
REGISTER sip:192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK24abfce3;rport
From: <sip:55432@192.168.1.111>;tag=as175817ba
To: <sip:55432@192.168.1.111>
Call-ID: 4a3e79ef47f427cd4540144e59c7aef7@192.168.1.111
CSeq: 1136 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="55432", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.111", nonce="49b8f910", response="78c089489190cc947f82cca393326c51"
Expires: 120
Contact: <sip:55432@192.168.1.112>
Event: registration
Content-Length: 0


---
trixbox1*CLI>
<--- SIP read from 192.168.1.111:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK24abfce3;received=192.168.1.112;rport=5060
From: <sip:55432@192.168.1.111>;tag=as175817ba
To: <sip:55432@192.168.1.111>
Call-ID: 4a3e79ef47f427cd4540144e59c7aef7@192.168.1.111
CSeq: 1136 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 192.168.1.111:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK24abfce3;received=192.168.1.112;rport=5060
From: <sip:55432@192.168.1.111>;tag=as175817ba
To: <sip:55432@192.168.1.111>;tag=as05df40ee
Call-ID: 4a3e79ef47f427cd4540144e59c7aef7@192.168.1.111
CSeq: 1136 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="302d1113"
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name 192.168.1.111
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.111:5060:
REGISTER sip:192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK3a019bb0;rport
From: <sip:55432@192.168.1.111>;tag=as04fdcc0a
To: <sip:55432@192.168.1.111>
Call-ID: 4a3e79ef47f427cd4540144e59c7aef7@192.168.1.111
CSeq: 1137 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="55432", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.111", nonce="302d1113", response="593ae8725c10be4b447503978a3897bd"
Expires: 120
Contact: <sip:55432@192.168.1.112>
Event: registration
Content-Length: 0


---
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.111:5060:
REGISTER sip:192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK534fb0db;rport
From: <sip:55432@192.168.1.111>;tag=as71ca26f5
To: <sip:55432@192.168.1.111>
Call-ID: 47c522423236c03631707e76060d93ef@192.168.1.111
CSeq: 1136 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="55432", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.111", nonce="3f7b7316", response="3308a2701a9b501c1ec4b3ce1c94071a"
Expires: 120
Contact: <sip:55432@192.168.1.112>
Event: registration
Content-Length: 0


---
trixbox1*CLI>
<--- SIP read from 192.168.1.111:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK3a019bb0;received=192.168.1.112;rport=5060
From: <sip:55432@192.168.1.111>;tag=as04fdcc0a
To: <sip:55432@192.168.1.111>
Call-ID: 4a3e79ef47f427cd4540144e59c7aef7@192.168.1.111
CSeq: 1137 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from 192.168.1.111:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK3a019bb0;received=192.168.1.112;rport=5060
From: <sip:55432@192.168.1.111>;tag=as04fdcc0a
To: <sip:55432@192.168.1.111>;tag=as05df40ee
Call-ID: 4a3e79ef47f427cd4540144e59c7aef7@192.168.1.111
CSeq: 1137 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 120
Contact: <sip:55432@192.168.1.112>;expires=120
Date: Fri, 30 Sep 2011 00:51:59 GMT
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Scheduling destruction of SIP dialog '4a3e79ef47f427cd4540144e59c7aef7@192.168.1.111' in 32000 ms (Method: REGISTER)
trixbox1*CLI>
<--- SIP read from 192.168.1.111:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK534fb0db;received=192.168.1.112;rport=5060
From: <sip:55432@192.168.1.111>;tag=as71ca26f5
To: <sip:55432@192.168.1.111>
Call-ID: 47c522423236c03631707e76060d93ef@192.168.1.111
CSeq: 1136 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from 192.168.1.111:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK534fb0db;received=192.168.1.112;rport=5060
From: <sip:55432@192.168.1.111>;tag=as71ca26f5
To: <sip:55432@192.168.1.111>;tag=as5adb06d6
Call-ID: 47c522423236c03631707e76060d93ef@192.168.1.111
CSeq: 1136 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5b949ec3"
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name 192.168.1.111
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.111:5060:
REGISTER sip:192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK4e502c52;rport
From: <sip:55432@192.168.1.111>;tag=as005655a2
To: <sip:55432@192.168.1.111>
Call-ID: 47c522423236c03631707e76060d93ef@192.168.1.111
CSeq: 1137 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="55432", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.111", nonce="5b949ec3", response="25a62b40cffad817b2d4e2c757569b6e"
Expires: 120
Contact: <sip:55432@192.168.1.112>
Event: registration
Content-Length: 0


---
trixbox1*CLI>
<--- SIP read from 192.168.1.111:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK4e502c52;received=192.168.1.112;rport=5060
From: <sip:55432@192.168.1.111>;tag=as005655a2
To: <sip:55432@192.168.1.111>
Call-ID: 47c522423236c03631707e76060d93ef@192.168.1.111
CSeq: 1137 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from 192.168.1.111:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK4e502c52;received=192.168.1.112;rport=5060
From: <sip:55432@192.168.1.111>;tag=as005655a2
To: <sip:55432@192.168.1.111>;tag=as5adb06d6
Call-ID: 47c522423236c03631707e76060d93ef@192.168.1.111
CSeq: 1137 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 120
Contact: <sip:55432@192.168.1.112>;expires=120
Date: Fri, 30 Sep 2011 00:51:59 GMT
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Scheduling destruction of SIP dialog '47c522423236c03631707e76060d93ef@192.168.1.111' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '5368afd0165f78f54bae898e1baef8c4@192.168.1.111' Method: OPTIONS
Really destroying SIP dialog '1953a3d55cae0cb05c933668227f9919@192.168.1.111' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.1.111:5060:
OPTIONS sip:22345@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK06e0cb49;rport
From: "Unknown" <sip:Unknown@192.168.1.112>;tag=as00ae7654
To: <sip:22345@192.168.1.111>
Contact: <sip:Unknown@192.168.1.112>
Call-ID: 0f3f6748606bf58c1e781d087e3ee9bb@192.168.1.112
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 30 Sep 2011 00:52:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
trixbox1*CLI>
<--- SIP read from 192.168.1.111:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK06e0cb49;received=192.168.1.112;rport=5060
From: "Unknown" <sip:Unknown@192.168.1.112>;tag=as00ae7654
To: <sip:22345@192.168.1.111>;tag=as4a7d46f6
Call-ID: 0f3f6748606bf58c1e781d087e3ee9bb@192.168.1.112
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:192.168.1.111>
Accept: application/sdp
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '0f3f6748606bf58c1e781d087e3ee9bb@192.168.1.112' Method: OPTIONS
Really destroying SIP dialog '4a3e79ef47f427cd4540144e59c7aef7@192.168.1.111' Method: REGISTER
Really destroying SIP dialog '47c522423236c03631707e76060d93ef@192.168.1.111' Method: REGISTER
apc
Silver Member
 
โพสต์: 33
ลงทะเบียนเมื่อ: 20 ก.ย. 2011 14:31
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Re: Got SIP response 489 "Bad event" back from แก้ยังไงครับ

โพสต์โดย nuiz » 30 ก.ย. 2011 09:25

อ่อ ได้หล่ะครับ ว่าเกิดจากอะไร
เกิดจาก .112 ส่ง NOTIFY message ไปหา .111 เพื่อบอกว่ามันมี voicemail ฝากไว้ แต่ว่า .111 มันไม่รองรับ NOTIFY message
แต่จากข้อความแล้ว ไม่มี Voicemail หรอก แต่มันเป็น Schedule ว่าถึงเวลาส่งก็ต้องส่ง ประมาณนี้ครับ ปัญหาไม่ได้อยู่ที่จำนวน Voicemail ว่ามีหรือไม่มี ปัญหาอยู่ที่ .111 ไม่รองรับ NOTIFY message หน่ะครับ

ดูจากข้อความที่ผมตัดมาให้ ตัดข้อความที่ไม่เกี่ยวข้องออกไป

---
Scheduling destruction of SIP dialog '56854c897b424cde144b78c56b40daa2@192.168.1.112' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.1.111:5060:
NOTIFY sip:22345@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK64820e8b;rport
From: "Unknown" <sip:Unknown@192.168.1.112>;tag=as083fe03d
To: <sip:22345@192.168.1.111>
Contact: <sip:Unknown@192.168.1.112>
Call-ID: 56854c897b424cde144b78c56b40daa2@192.168.1.112
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*97@192.168.1.112
Voice-Message: 0/0 (0/0)


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '0f24e85d6b583ddd7f1209ca32f72cfa@192.168.1.112' Method: OPTIONS
trixbox1*CLI>
<--- SIP read from 192.168.1.111:5060 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 192.168.1.112:5060;branch=z9hG4bK64820e8b;received=192.168.1.112;rport=5060
From: "Unknown" <sip:Unknown@192.168.1.112>;tag=as083fe03d
To: <sip:22345@192.168.1.111>;tag=as19764809
Call-ID: 56854c897b424cde144b78c56b40daa2@192.168.1.112
CSeq: 102 NOTIFY
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
-- Got SIP response 489 "Bad event" back from 192.168.1.111



และเป็นอะไรที่เกี่ยวกับการแจ้งเตือนเฉยๆ ฉะนั้นจึงไม่มีผลต่อการโทรครับ สบายใจได้

ถ้าจะปิดไม่ให้มี Event แบบนี้ คงต้องไปเช็คที่ .111 ว่าทำยังไงจะให้มันรองรับ NOTIFY หรือไม่ก็มาเช็คที่ .112 ว่าทำยังไงจะไม่ให้มันส่ง NOTIFY ออกไปหา .111 ก็คงจะโดยการปิด Voicemail ของเบอร์ Extension ของ .111 หน่ะครับ

ประมาณนี้ครับ
** หากมีปัญหากับอุปกรณ์ที่ซื้อมาเองหรือบริการที่ทำขึ้นมาเอง ให้โพสต์ถามในเว็บบอร์ดนี้นะครับ **
** งานเร่งด่วนติดต่อว่าจ้างที่เบอร์ 08-5161-9439 อีเมล์ iamaladin@gmail.com ไลน์ NuizVoip ครับ **
nuiz
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ลงทะเบียนเมื่อ: 24 มี.ค. 2010 09:33
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Re: Got SIP response 489 "Bad event" back from แก้ยังไงครับ

โพสต์โดย apc » 30 ก.ย. 2011 11:37

สุดยอดไปเลยครับ ขอบคุณมากๆเลยครับ
apc
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ลงทะเบียนเมื่อ: 20 ก.ย. 2011 14:31
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Re: Got SIP response 489 "Bad event" back from แก้ยังไงครับ

โพสต์โดย apc » 03 ต.ค. 2011 09:40

ยังหาที่ปิดไม่ได้เลยครับ :roll:
apc
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ลงทะเบียนเมื่อ: 20 ก.ย. 2011 14:31
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