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Bummer - we've got an office full of these phones :(<br>
<br>
As you suggest, I'll turn off all the NAT stuff when I'm there on
Monday and see if that fixes it - hope so, otherwise I guess it
looks like I'll have to try and get Aastra to look at the problem
(doubt they are going to be very interested though!).<br>
<br>
Thanks,<br>
Charlie<br>
<br>
<br>
On 03/12/2011 06:29, Joegen Baclor wrote:
<blockquote cite="mid:4ED9B3A8.4040707@ezuce.com" type="cite">
<meta content="text/html; charset=ISO-8859-1"
http-equiv="Content-Type">
It's definitely a bug on the phone. I would try setting all
Aastra NAT related features off and see if it stops getting
confused like this. Freeswitch should be able to handle NAT from
the far end. <br>
<br>
On 12/02/2011 08:18 PM, Charlie Orford wrote:
<blockquote cite="mid:4ED8C206.5010502@attackplan.net" type="cite">
<meta http-equiv="Context-Type" content="text/html;
charset=ISO-8859-1">
Thanks Joegen, I hadn't picked up on that.<br>
<br>
Is there any valid reason why the phone would swap to using it's
internal IP (192.168.0.175) for the Via and Contact headers when
responding to the auth challenge? Or does this look more like a
bug with the phone?<br>
<br>
<br>
<br>
On 02/12/2011 12:50, Joegen Baclor wrote:
<blockquote cite="mid:4ED8BB73.8080401@ezuce.com" type="cite">
For some reason, the aastra phone is sending a different
contact and via address after it gets challenged by
freeswitch.<br>
<br>
Original address was <span><br>
<br>
</span> <span>Via: SIP/</span><span>2.0</span><span>/UDP<span> </span></span><span>7.7</span><span>.7</span><span>.7</span><span>:</span><span>6060</span><span>;branch=z9hG4bK2ff03b684c51c6e9a.84815b35f84df80f4;rport</span>
(LINE# 35)<br>
<br>
Then it changes to <br>
<br>
<span>Via: SIP/</span><span>2.0</span><span>/UDP<span> </span></span><span>192.168</span><span>.0</span><span>.175</span><span>:</span><span>6060</span><span>;branch=z9hG4bK085dcaff3e1ae346b.e0c0723efae179659;rport</span>
(LINE# 187)<br>
<br>
It then sends a CANCEL using <span>Via: SIP/</span><span>2.0</span><span>/UDP<span> </span></span><span>7.7</span><span>.7</span><span>.7</span><span>:</span><span>6060
(LINE# 961) which freeswitch isn't able to match to a
transaction because it is probably expecting </span><span><span></span></span><span>192.168</span><span>.0</span><span>.175.<br>
<br>
<br>
</span><br>
On 12/02/2011 07:19 PM, Charlie Orford wrote:
<blockquote cite="mid:4ED8B456.9000505@attackplan.net"
type="cite"> Hi Anthony<br>
<br>
We are using Aastra 57i phones with the latest firmware
(v3.2.2.56 from June 2011).<br>
<br>
However, it looks like a specific issue with the Aastra 57i
as I have just replicated the call and cancel situation
using a 3CX softphone and in this case the CANCEL request
gets honoured by FS.<br>
<br>
Here's a copy of a failed CANCEL test using the Aastra
phone: <a moz-do-not-send="true"
href="http://pastebin.freeswitch.org/17922">http://pastebin.freeswitch.org/17922</a><br>
<br>
Here's a copy of a successful CANCEL test using the 3CX
softphone: <a moz-do-not-send="true"
href="http://pastebin.freeswitch.org/17923">http://pastebin.freeswitch.org/17923</a><br>
<br>
sofia loglevel all 9 is on for both tests.<br>
<br>
Charlie<br>
<br>
<br>
On 01/12/2011 21:53, Anthony Minessale wrote:
<blockquote
cite="mid:CAKbxfG_UrV0q=+Pu_ja+kftrvuuB-FN3oKiG6H08_uUpDr2CGQ@mail.gmail.com"
type="cite">try same failed call test with sofia loglevel
all 9
<div><br>
</div>
<div>Also, try some other phone or device that does not
have the problem, and create a cancel situation the same
way and see if you can find a difference.</div>
<div><br>
</div>
<div>Finally, have you tried the latest firmware on the
phone?</div>
<div><br>
<br>
<div>On Thu, Dec 1, 2011 at 10:17 AM, Charlie Orford <span><<a
moz-do-not-send="true"
href="mailto:charlie.orford@attackplan.net">charlie.orford@attackplan.net</a>></span>
wrote:<br>
<blockquote>Hi list,<br>
<br>
When we make a call from an FS extension to a PSTN
number (via our ITSP<br>
gateway provider) and hang-up before the call
completes, FS replies to<br>
the CANCEL request with "481 Call/Transaction Does
Not Exist" and the<br>
call continues to ring on the remote end. If we
hangup after the call<br>
has connected, it works with no problem.<br>
<br>
I have looked through a SIP trace of this happening
and to my (untrained<br>
eye) nothing seems obviously wrong (i.e. tag, call
id and branch values<br>
all seem to be correct). I'm using the latest git
snapshot from<br>
2011-11-30 18-14-24 -0600.<br>
<br>
For a sip trace showing the problem, see:<br>
<a moz-do-not-send="true"
href="http://pastebin.freeswitch.org/17908">http://pastebin.freeswitch.org/17908</a><br>
<br>
For a sip trace showing a successful hangup, see:<br>
<a moz-do-not-send="true"
href="http://pastebin.freeswitch.org/17909">http://pastebin.freeswitch.org/17909</a><br>
<br>
*Note: ip addresses, domain and called number have
been altered for privacy.<br>
<br>
Any help or insight is much appreciated.<br>
<br>
Kind Regards,<br>
Charlie<br>
<br>
<br>
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-- <br>
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<br>
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