Softverkpro
asked on
Trixbox: SIP/2.0 489 Bad event Error on incoming calls
hi,
i have this really wierd and annoying error. we have obtained a sip trunk from Siminn Iceland Ltd provider and set it up with trixbox. the trunk can register with the service provider just fine. i can set the trunk as a peer/friend ( type = peer or type = friend and setup incoming context and type = user) i am able to make outbound calls through it. but as soon as i set it up like that all the incoming call is rejected by asterisk and from the console i can see it is a "SIP/2.0 489 Bad event" msg generated by asterisk. if i get rid of all the "type" command from the trunk setup, i can receive calls just as fine. but obviously that stops me from calling outbound through the trunk.
the configuration for receiving making outgoing call is like this:
[siminn]
host=157.157.16.140
username=2000053544990160
secret=somePassword
user=phone
type=friend
register string:
2000053544990160: somePassword@157.157.16.14 0/20000535 44990160
in this configuration if i receive an incoming call, this is what i see on the asterisk console:
trixbox1*CLI>
<--- SIP read from 157.157.16.140:5060 --->
INVITE sip:2000053544990160@202.1 76.88.148 SIP/2.0
Via: SIP/2.0/UDP 157.157.16.140:5060;branch =z9hG4bK49 12b162-858 7-5936
Call-Id: CI_430507_2000053544990160 @1eaaaaaaa 271_404041
From: "" ;tag=157.157.16.140-8587-1 495
To:
Max-Forwards: 70
CSeq: 430507 INVITE
Contact:
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 157.157.16.140 : 5060 (no NAT)
Using INVITE request as basis request - CI_430507_2000053544990160 @1eaaaaaaa 271_404041
Found peer 'simin'
trixbox1*CLI>
<--- Reliably Transmitting (no NAT) to 157.157.16.140:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 157.157.16.140:5060;branch =z9hG4bK49 12b162-858 7-5936;rec eived=157. 157.16.140
From: "" ;tag=157.157.16.140-8587-1 495
To: ;tag=as5c42a171
Call-ID: CI_430507_2000053544990160 @1eaaaaaaa 271_404041
CSeq: 430507 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39194225"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'CI_430507_200005354499016 0@1eaaaaaa a271_40404 1' in 32000 ms (Method: INVITE)
trixbox1*CLI>
<--- SIP read from 157.157.16.140:5060 --->
ACK sip:2000053544990160@202.1 76.88.148 SIP/2.0
Via: SIP/2.0/UDP 157.157.16.140:5060;branch =z9hG4bK49 12b162-858 7-5936
Call-Id: CI_430507_2000053544990160 @1eaaaaaaa 271_404041
From: "" ;tag=157.157.16.140-8587-1 495
To: ;tag=as5c42a171
CSeq: 430507 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from 157.157.16.140:5060 --->
NOTIFY sip:2000053544990160@202.1 76.88.148 SIP/2.0
Via: SIP/2.0/UDP 157.157.16.140:5060;branch =z9hG4bK49 12b162-903 2-9620
Call-Id: CI_291719_2000053544990160 @1eaaaaaaa 271_274688
From: sip:2000053544990160@194.1 44.220.247 :1105;tag= 157.157.16 .140-42949 65200-8974
To: sip:2000053544990160@194.1 44.220.247 :1105
Max-Forwards: 70
CSeq: 219 NOTIFY
Contact:
Subscription-State: active
Content-Type: application/simple-message -summary
Event: message-summary
Content-Length: 42
Messages-Waiting: no
Voice-Message: 0/0
<------------->
--- (12 headers 2 lines) ---
trixbox1*CLI>
<--- Transmitting (no NAT) to 157.157.16.140:5060 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 157.157.16.140:5060;branch =z9hG4bK49 12b162-903 2-9620;rec eived=157. 157.16.140
From: sip:2000053544990160@194.1 44.220.247 :1105;tag= 157.157.16 .140-42949 65200-8974
To: sip:2000053544990160@194.1 44.220.247 :1105;tag= as3b0ed1c9
Call-ID: CI_291719_2000053544990160 @1eaaaaaaa 271_274688
CSeq: 219 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
ontent-Length: 0
but if i change the configuration to the following, i can receive calls, but it will stop me from making outbound calls because no peer is defined:
[siminn]
host=157.157.16.140
username=2000053544990160
secret=somePassword
user=phone
register string:
2000053544990160: somePassword@157.157.16.14 0/20000535 44990160
i have also tried splitting it into two parts (type = peer and type = user). but did not help.
anyone have any idea?
khan
i have this really wierd and annoying error. we have obtained a sip trunk from Siminn Iceland Ltd provider and set it up with trixbox. the trunk can register with the service provider just fine. i can set the trunk as a peer/friend ( type = peer or type = friend and setup incoming context and type = user) i am able to make outbound calls through it. but as soon as i set it up like that all the incoming call is rejected by asterisk and from the console i can see it is a "SIP/2.0 489 Bad event" msg generated by asterisk. if i get rid of all the "type" command from the trunk setup, i can receive calls just as fine. but obviously that stops me from calling outbound through the trunk.
the configuration for receiving making outgoing call is like this:
[siminn]
host=157.157.16.140
username=2000053544990160
secret=somePassword
user=phone
type=friend
register string:
2000053544990160: somePassword@157.157.16.14
in this configuration if i receive an incoming call, this is what i see on the asterisk console:
trixbox1*CLI>
<--- SIP read from 157.157.16.140:5060 --->
INVITE sip:2000053544990160@202.1
Via: SIP/2.0/UDP 157.157.16.140:5060;branch
Call-Id: CI_430507_2000053544990160
From: "" ;tag=157.157.16.140-8587-1
To:
Max-Forwards: 70
CSeq: 430507 INVITE
Contact:
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 157.157.16.140 : 5060 (no NAT)
Using INVITE request as basis request - CI_430507_2000053544990160
Found peer 'simin'
trixbox1*CLI>
<--- Reliably Transmitting (no NAT) to 157.157.16.140:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 157.157.16.140:5060;branch
From: "" ;tag=157.157.16.140-8587-1
To: ;tag=as5c42a171
Call-ID: CI_430507_2000053544990160
CSeq: 430507 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39194225"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'CI_430507_200005354499016
trixbox1*CLI>
<--- SIP read from 157.157.16.140:5060 --->
ACK sip:2000053544990160@202.1
Via: SIP/2.0/UDP 157.157.16.140:5060;branch
Call-Id: CI_430507_2000053544990160
From: "" ;tag=157.157.16.140-8587-1
To: ;tag=as5c42a171
CSeq: 430507 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from 157.157.16.140:5060 --->
NOTIFY sip:2000053544990160@202.1
Via: SIP/2.0/UDP 157.157.16.140:5060;branch
Call-Id: CI_291719_2000053544990160
From: sip:2000053544990160@194.1
To: sip:2000053544990160@194.1
Max-Forwards: 70
CSeq: 219 NOTIFY
Contact:
Subscription-State: active
Content-Type: application/simple-message
Event: message-summary
Content-Length: 42
Messages-Waiting: no
Voice-Message: 0/0
<------------->
--- (12 headers 2 lines) ---
trixbox1*CLI>
<--- Transmitting (no NAT) to 157.157.16.140:5060 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 157.157.16.140:5060;branch
From: sip:2000053544990160@194.1
To: sip:2000053544990160@194.1
Call-ID: CI_291719_2000053544990160
CSeq: 219 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
ontent-Length: 0
but if i change the configuration to the following, i can receive calls, but it will stop me from making outbound calls because no peer is defined:
[siminn]
host=157.157.16.140
username=2000053544990160
secret=somePassword
user=phone
register string:
2000053544990160: somePassword@157.157.16.14
i have also tried splitting it into two parts (type = peer and type = user). but did not help.
anyone have any idea?
khan
Did you check the codec of the Inbound call and if your trixbox supports such codec, even if the extension support it?
Regards
Regards
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ASKER
sorry the question looks a bit malformed here. i have also asked for solution in trixbox forum. if you have trouble reading iit here, please check out the same problem here:
http://trixbox.org/forums/trixbox-forums/trunks/sip-2-0-489-bad-event-error-incoming-calls