Hi James,<div><br></div><div>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,</div><div><br>
</div><div><div><-- SIP read from <a href="http://98.158.181.173:5060">98.158.181.173:5060</a>:</div><div>NOTIFY <a href="http://sip:PBXFamilia@10.0.8.84:5060">sip:PBXFamilia@10.0.8.84:5060</a> SIP/2.0</div><div>Via: SIP/2.0/UDP 98.158.181.173:5060;branch=z9hG4bK47ff44c5;rport</div>
<div>From: "Unknown" <<a href="mailto:sip%3AUnknown@98.158.181.173">sip:Unknown@98.158.181.173</a>>;tag=as24f09d54</div><div>To: <<a href="http://sip:PBX@10.0.8.84:5060">sip:PBX@10.0.8.84:5060</a>></div>
<div>Contact: <<a href="mailto:sip%3AUnknown@98.158.181.173">sip:Unknown@98.158.181.173</a>></div><div>Call-ID: <a href="mailto:03d488a828e9bee61ca72fc16f3789b7@98.158.181.173">03d488a828e9bee61ca72fc16f3789b7@98.158.181.173</a></div>
<div>CSeq: 102 NOTIFY</div><div>User-Agent: Asterisk PBX</div><div>Max-Forwards: 70</div><div>Event: message-summary</div><div>Content-Type: application/simple-message-summary</div><div>Content-Length: 89</div><div><br></div>
<div>Messages-Waiting: no</div><div>Message-Account: sip:*<a href="mailto:97@98.158.181.173">97@98.158.181.173</a></div><div>Voice-Message: 0/0 (0/0)</div><div><br></div><div>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)---</div>
<div>Oct 6 08:57:43 VERBOSE[31038] logger.c: Trasmitting Response: 489 Bad event</div><div>Oct 6 08:57:43 VERBOSE[31038] logger.c: HERE chan_sip.c ast_sip_ouraddrfor 1365</div><div>Oct 6 08:57:43 VERBOSE[31038] logger.c: Transmitting (no NAT) to <a href="http://98.158.181.173:5060">98.158.181.173:5060</a>:</div>
<div>SIP/2.0 489 Bad event</div><div>Via: SIP/2.0/UDP 98.158.181.173:5060;branch=z9hG4bK47ff44c5;rport;received=98.158.181.173</div><div>From: "Unknown" <<a href="mailto:sip%3AUnknown@98.158.181.173">sip:Unknown@98.158.181.173</a>>;tag=as24f09d54</div>
<div>To: <<a href="http://sip:PBXFamilia@10.0.8.84:5060">sip:PBXFamilia@10.0.8.84:5060</a>>;tag=as3cdb539f</div><div>Call-ID: <a href="mailto:03d488a828e9bee61ca72fc16f3789b7@98.158.181.173">03d488a828e9bee61ca72fc16f3789b7@98.158.181.173</a></div>
<div>CSeq: 102 NOTIFY</div><div>User-Agent: CEM PBX</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</div><div>Max-Forwards: 70</div><div>Content-Length: 0</div><div><br></div><div>My Setup: I have created one extension in elastix and the extension is configured as VoIP trunk in Asterisk. </div>
<div><br></div><div><br></div><div><br></div><br><div class="gmail_quote">On Sun, Apr 11, 2010 at 1:10 PM, Adrian Marsh <span dir="ltr"><<a href="mailto:Adrian.Marsh@ubiquisys.com">Adrian.Marsh@ubiquisys.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hi James,<br>
<br>
Thanks for the help. 3.10 registers into my SIP server just as a normal SIP client.<br>
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<br>
<font color="#888888"><br>
Adrian<br>
</font><div><div></div><div class="h5"><br>
> Hi All,<br>
><br>
><br>
><br>
> I've two asterisk servers on the same LAN, both 1.4, and I keep getting "Got<br>
> SIP response 489 "Bad event" back from 192.168.3.10"<br>
><br>
> No idea whats causing it. The only references I can find mentions NATing<br>
> issues, but these are on the same LAN so NAT shouldn't be an issue.<br>
><br>
> 3.10 does authenticate into the server logging the error. The error appears<br>
> in the log every 1m20s (ish)<br>
<br>
Is 3.10 on a SIP trunk to the other asterisk box?<br>
Is qualify=yes on this SIP trunk?<br>
I think you'll find that if you run an ngrep/tcpdump on port 5060 on<br>
the box receiving the error it will send out an OPTIONS or NOTIFY (I<br>
can't remember which) and then you'll see the 489 Bad Event.<br>
Grab a trace of the SIP traffic and post it, its the only way to know<br>
for sure though.<br>
<br>
-- James<br>
<br>
><br>
><br>
><br>
> Any ideas?<br>
><br>
><br>
><br>
> Thanks,<br>
><br>
><br>
><br>
> Adrian<br>
><br>
><br>
><br>
> --<br>
> _____________________________________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
><br>
> asterisk-users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
><br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
<br>
</div></div>--<br>
<div><div></div><div class="h5">_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div><br><br clear="all"><br>-- <br>Thank you with regards,<br>Gops<br><br><br>
</div>