<div dir="ltr"><br><div class="gmail_extra"><br><div class="gmail_quote">On Tue, May 21, 2013 at 11:26 AM, Matthew J. Roth <span dir="ltr"><<a href="mailto:mroth@imminc.com" target="_blank">mroth@imminc.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div class="im">asterisk users wrote:<br>
><br>
> I'm having a strange problem recently with a Yealink SIP-T28P phone connected<br>
> to Asterisk 11.4.0 via openvpn. It was working fine for months, and now when I<br>
> dial anything from the phone, it shows "Forbidden", and the Asterisk console<br>
> shows:<br>
><br>
> [May 21 10:47:49] NOTICE[28518][C-00000004]: chan_sip.c:25189 handle_request_invite: Failed to authenticate device "Ext 110" < <a href="mailto:sip%3A110@192.168.6.2">sip:110@192.168.6.2</a> >;tag=1130259112<br>
><br>
> Asterisk 192.168.6.2<br>
> OpenVPN on router 10.8.0.1<br>
> Remote Yealink phone 10.8.0.6<br>
><br>
> The remote phone shows as being registered:<br>
> PBX*CLI> sip show peers<br>
> Name/username Host Dyn Forcerport ACL Port Status Description<br>
> 110/110 10.8.0.6 D A 5062 OK (111 ms) Yealink OpenVPN<br>
><br>
> Also, if there is voicemail in the mailbox for 110, the phone's message light<br>
> is lit and it beeps periodically.<br>
><br>
</div>> ...<br>
<div class="im">><br>
> Any suggestions on what might be happening here, and how it could be resolved?<br>
<br>
<br>
</div>That is quite strange. Please provide SIP traces of the dialogs between<br>
Asterisk and the phone in the following two scenarios:<br>
<br>
1) Phone registering to Asterisk (presumably successful)<br>
2) Phone dialing to Asterisk (presumably unsuccessful)<br>
<br>
Regards,<br>
<br>
Matthew Roth<br>
InterMedia Marketing Solutions<br>
Software Engineer and Systems Developer<br>
<br>
--<br>
<br></blockquote><div><br><div>Registration trace<br></div>(note that extension 88 is the voicemail extension, which the phone registers to also for MWI)<br><div>--> <a href="http://pastebin.com/c3H700wa">http://pastebin.com/c3H700wa</a><br>
<br></div><div>Call trace:<br>|Time | 10.8.0.6 |<br>| | | 192.168.6.2 | <br>|268.693661|
INVITE SDP (g711U g729 g722 telephone-eventRTP...e-101) |SIP
From: "Ext 110" <<a href="mailto:sip%3A110@192.168.6.2">sip:110@192.168.6.2</a> To:<<a href="mailto:sip%3A88@192.168.6.2">sip:88@192.168.6.2</a><br>| |(1024) ------------------> (5060) |<br>
|268.694449| 401 Unauthorized |SIP Status<br>| |(1024) <------------------ (5060) |<br>|268.914195| ACK | |SIP Request<br>| |(1024) ------------------> (5060) |<br>
|268.945115|
INVITE SDP (g711U g729 g722 telephone-eventRTP...e-101) |SIP
From: "Ext 110" <<a href="mailto:sip%3A110@192.168.6.2">sip:110@192.168.6.2</a> To:<<a href="mailto:sip%3A88@192.168.6.2">sip:88@192.168.6.2</a><br>| |(1024) ------------------> (5060) |<br>
|268.945717| 403 Forbidden |SIP Status<br>| |(1024) <------------------ (5060) |<br>|269.041417| ACK | |SIP Request<br>| |(1024) ------------------> (5060) |<br>
<br><br></div>I'm also confused by the reference in "sip show peers" to port 5062, as I can't see that anywhere in the configuration of either the phone or in sip.conf. All the other phones show port 5060 in the "sip show peers" output.<br>
</div><div> </div></div><br></div></div>