Hi James, I too facing the same issue whereas in the inbound call I am able to receive the call, when I pickup the receiver it hangsup. I am getting the NOTIFY option.. the log as follows,
<-- SIP read from 98.158.181.173:5060: NOTIFY sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 98.158.181.173:5060;branch=z9hG4bK47ff44c5;rport From: "Unknown" <sip:[email protected] <sip%[email protected]> >;tag=as24f09d54 To: <sip:[email protected]:5060> Contact: <sip:[email protected] <sip%[email protected]>> Call-ID: [email protected] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89 Messages-Waiting: no Message-Account: sip:*[email protected] Voice-Message: 0/0 (0/0) Oct 6 08:57:43 VERBOSE[31038] logger.c: --- (12 headers 3 lines)Oct 6 08:57:43 VERBOSE[31038] logger.c: --- (12 headers 3 lines)--- Oct 6 08:57:43 VERBOSE[31038] logger.c: Trasmitting Response: 489 Bad event Oct 6 08:57:43 VERBOSE[31038] logger.c: HERE chan_sip.c ast_sip_ouraddrfor 1365 Oct 6 08:57:43 VERBOSE[31038] logger.c: Transmitting (no NAT) to 98.158.181.173:5060: SIP/2.0 489 Bad event Via: SIP/2.0/UDP 98.158.181.173:5060 ;branch=z9hG4bK47ff44c5;rport;received=98.158.181.173 From: "Unknown" <sip:[email protected] <sip%[email protected]> >;tag=as24f09d54 To: <sip:[email protected]:5060>;tag=as3cdb539f Call-ID: [email protected] CSeq: 102 NOTIFY User-Agent: CEM PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Content-Length: 0 My Setup: I have created one extension in elastix and the extension is configured as VoIP trunk in Asterisk. On Sun, Apr 11, 2010 at 1:10 PM, Adrian Marsh <[email protected]>wrote: > Hi James, > > Thanks for the help. 3.10 registers into my SIP server just as a normal > SIP client. > Yes, qualify=yes. I just tried setting that to no on my end, and I still > get the message. I'll try turning it off on 3.10 too tomorrow and capture > some trace too > > Adrian > > > Hi All, > > > > > > > > I've two asterisk servers on the same LAN, both 1.4, and I keep getting > "Got > > SIP response 489 "Bad event" back from 192.168.3.10" > > > > No idea whats causing it. The only references I can find mentions NATing > > issues, but these are on the same LAN so NAT shouldn't be an issue. > > > > 3.10 does authenticate into the server logging the error. The error > appears > > in the log every 1m20s (ish) > > Is 3.10 on a SIP trunk to the other asterisk box? > Is qualify=yes on this SIP trunk? > I think you'll find that if you run an ngrep/tcpdump on port 5060 on > the box receiving the error it will send out an OPTIONS or NOTIFY (I > can't remember which) and then you'll see the 489 Bad Event. > Grab a trace of the SIP traffic and post it, its the only way to know > for sure though. > > -- James > > > > > > > > > Any ideas? > > > > > > > > Thanks, > > > > > > > > Adrian > > > > > > > > -- > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Thank you with regards, Gops
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
