489 Bad Event

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489 Bad Event

Postby CustomGT » Thu Dec 03, 2009 9:17 pm

66.135.32.160 = asterisk server
70.238.155.70 = ip of the public network that the client phone is in
10.0.0.34 = client phone ip

Im not sure why it is saying 489 Bad Event...any ideas?

<------------->
--- (13 headers 0 lines) ---
Scheduling destruction of SIP dialog '47911e8036aeb26a7452867871cb0ee8@66.135.32.160' in 32000 ms (Method: REGISTER)
pbx1*CLI>
<--- SIP read from 70.238.155.70:5060 --->
NOTIFY sip:pbx1.myserver.com SIP/2.0
Via: SIP/2.0/UDP 10.0.0.34:5060;branch=z9hG4bK-b6118ff0
From: 1235 <sip:1235@pbx1.myserver.com>;tag=9a83ed6654e8b54do1
To: <sip:pbx1.myserver.com>
Call-ID: cb99f356-b1a0e81d@10.0.0.34
CSeq: 117 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA2100-3.2.2(g)
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 10.0.0.34 : 5060 (no NAT)
pbx1*CLI>
<--- Transmitting (no NAT) to 10.0.0.34:5060 --->
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 10.0.0.34:5060;branch=z9hG4bK-b6118ff0;received=70.238.155.70
From: 1235 <sip:1235@pbx1.myserver.com>;tag=9a83ed6654e8b54do1
To: <sip:pbx1.myserver.com>;tag=as032b74fc
Call-ID: cb99f356-b1a0e81d@10.0.0.34
CSeq: 117 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
CustomGT
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Posts: 224
Joined: Thu Dec 21, 2006 10:56 pm

Re: 489 Bad Event

Postby david55 » Fri Dec 04, 2009 6:20 am

Because it doesn't know what "keep-alive" means. About the only type of event it supports is "refer".
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Re: 489 Bad Event

Postby CustomGT » Fri Dec 04, 2009 8:38 am

david55 wrote:Because it doesn't know what "keep-alive" means. About the only type of event it supports is "refer".


Ok, so your saying the only events that this asterisk supports is?:
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

And keep-alive is not one of them, which this ATA is trying to send to asterisk.

I seen this was an issue with this ATA. Im on 1.4 so no fixing it for me:
https://issues.asterisk.org/view.php?id=15084


Thank you for your help, this helps me more than you know
CustomGT
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Posts: 224
Joined: Thu Dec 21, 2006 10:56 pm

Re: 489 Bad Event

Postby david55 » Fri Dec 04, 2009 8:45 am

This is one of those, NOTIFY!

What I'm actually saying is that Asterisk (source code for 1.6.1.0) only recognizes Refer events in NOTIFY requests. It shouldn't be getting others, because it will not have subscribed to them. You need to stop your equipment issuing unsolicited NOTIFYs, if you don't want Asterisk to reject them.

A quick Google suggests to me that this event is not standardised in any RFC, although I could be wrong.
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Re: 489 Bad Event

Postby CustomGT » Fri Dec 04, 2009 9:10 am

david55 wrote:This is one of those, NOTIFY!

What I'm actually saying is that Asterisk (source code for 1.6.1.0) only recognizes Refer events in NOTIFY requests. It shouldn't be getting others, because it will not have subscribed to them. You need to stop your equipment issuing unsolicited NOTIFYs, if you don't want Asterisk to reject them.

A quick Google suggests to me that this event is not standardised in any RFC, although I could be wrong.


Ok that makes sense! Here is how i fixed it in case anyone needs help in the future:
http://www.voip-info.org/wiki/view/Sipu ... ion+Line+1
This talks about disabling it on the ATA.
In the admin, then advanced, than Line tab, NAT Keep Alive Enable: NO

David, one last thing on these sip dialogs, im getting an options now. Ive read thru the openser book on sip, and read thru the RFC on sip...from what i can tell the client is asking asterisk what the OPTIONS are, but im not sure what options it would want to know if there is no call being made. They are just sitting idle...what OPTIONS are they talking about?:

Reliably Transmitting (NAT) to 70.238.155.70:5060:
OPTIONS sip:1235@70.238.155.70:5060 SIP/2.0
Via: SIP/2.0/UDP 66.135.32.160:5060;branch=z9hG4bK22cbb87b;rport
From: "Unknown" <sip:Unknown@66.135.32.160>;tag=as7c9d0dbc
To: <sip:1235@70.238.155.70:5060>
Contact: <sip:Unknown@66.135.32.160>
Call-ID: 67ed69ce5bd27d9162a765ee0adc09ed@66.135.32.160
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 04 Dec 2009 15:03:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
pbx1*CLI>
<--- SIP read from 70.238.155.70:5060 --->
SIP/2.0 200 OK
To: <sip:1235@70.238.155.70:5060>;tag=7ed488c75dd68cai1
From: "Unknown" <sip:Unknown@66.135.32.160>;tag=as7c9d0dbc
Call-ID: 67ed69ce5bd27d9162a765ee0adc09ed@66.135.32.160
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 66.135.32.160:5060;branch=z9hG4bK22cbb87b
Server: Sipura/SPA2100-3.2.2(g)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
CustomGT
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Posts: 224
Joined: Thu Dec 21, 2006 10:56 pm

Re: 489 Bad Event

Postby david55 » Fri Dec 04, 2009 9:12 am

You have "qualify" enabled. Asterisk is testing whether the client is still there. (I.E. it is Asterisk's way of doing keep alive.)
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Re: 489 Bad Event

Postby CustomGT » Fri Dec 04, 2009 9:28 am

david55 wrote:You have "qualify" enabled. Asterisk is testing whether the client is still there. (I.E. it is Asterisk's way of doing keep alive.)


102 OPTIONS is a wierd way of saying QUALIFY...
Thanks
CustomGT
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Posts: 224
Joined: Thu Dec 21, 2006 10:56 pm


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