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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Is your Cisco device a Cisco router if so
make sure you have no sip fixup.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>The Cisco may be fudging the SIP headers.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Alex<o:p></o:p></span></font></p>
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10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>AL Daei<br>
<b><span style='font-weight:bold'>Sent:</span></b> Sunday, January 14, 2007
11:57 AM<br>
<b><span style='font-weight:bold'>To:</span></b>
asterisk-users@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> [asterisk-users] RE
polycom fails registration</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'>nat is equal to yes,<br>
and server definition in ftp provisioning server is correct.<br>
i followed packets between phone and asterisk,<br>
it seems for some reason asterisk is not happy about challenge response its
getting from polycom.<br>
why its not happening in the same LAN, beats me!<br>
and also NAT device is cisco SIP aware and works flawlessly with
Linksys.<br>
the only close definition of issue i have found is in wiki as i mentioned.<br>
for reminder here it is:<br>
"If the phones fail to register with Asterisk but can still make outbound
calls, you likely need to adjust the digest realm parameter from the default of
PolycomSPIP."<br>
i'll try to use SIP image for polycom and let you guys know.<br>
<confused!><br>
<br>
>I have never seen a registration failure solved with nat=yes.<br>
<br>
Doug Lytle wrote:<br>
> Al wrote:<br>
>> I'm facing a weird issue, polycom phones work fine in the main office,
<br>
>> in remote office it says,<br>
>> Registration from '<sip:<a href="mailto:202@10.0.1.190">202@10.0.1.190</a>>'
failed for '70.59.21.112' - <br>
>> Wrong password<br>
>> the odd thing is Linksys phone works without any issue!!<br>
> <br>
> Just a guess, put nat=yes in the sip.conf for that phone and see if it <br>
> helps.<br>
> <br>
> Don't forget to restart Asterisk<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma'>Try amazing new 3D maps <a
href="http://maps.live.com/?wip=51" target="_new">Check it out!</a><o:p></o:p></span></font></p>
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