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<p class=MsoNormal><span style='color:#1F497D'>I am using an Edgwater device
instead of a Ingate Siparator. Where can I find all the Customer Parameters for
a Sip Profile on ShoreTel?<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Jason L. Nesheim
[mailto:jnesheim@cytek.biz] <br>
<b>Sent:</b> Monday, October 19, 2009 5:45 PM<br>
<b>To:</b> Ujjval Karihaloo<br>
<b>Cc:</b> VoiceOps@voiceops.org<br>
<b>Subject:</b> Re: [VoiceOps] Shoretel 9.0 Call transfer using SIP softphone
Xlite/Eyebeam<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='color:black'>You
have two options that I know of, one which may or may not work depending on
your network.<br>
<br>
You can add an Ingate Firewall/Siparator to the deployment. This device
will locally handle the REFER operation and actually send the call back to the
ShoreTel PBX (because of the Refer-To target) which then routes it through the
appropriate trunk on the PBX stripping the access code before it sends it back
to the Ingate. This also avoids the SDPless Invite on transfer issue
below.<br>
<br>
You can also try manually disabling the use of REFER as the transfer
method. To disable REFER you can edit the SIP profile for your trunk and
set "DontFwdREFER=1" under the custom parameters box. Be aware
that this may cause other problems with call transfers and three way
calling. I've been unable to get the PBX to work correctly on SIP trunks
without the Ingate when REFER is disabled and G729 is used. This is due
to the ShoreTel sending SDPless Invites when starting three way conferences and
then subsequently failing to handle a 200 OK response with G729a in the offer
list (it simply drops the messages). If you restrict the list of codecs
in the SDP response to the Invite to G711u only it should work for you.<o:p></o:p></span></p>
<div>
<div>
<div>
<p class=MsoNormal><span style='color:black'>-- <br>
Jason Nesheim<o:p></o:p></span></p>
</div>
</div>
</div>
<p class=MsoNormal><span style='color:black'><br>
<br>
----- Original Message -----<br>
From: "Ujjval Karihaloo" <ujjval@simplesignal.com><br>
To: "Jason L. Nesheim" <jnesheim@cytek.biz><br>
Cc: VoiceOps@voiceops.org<br>
Sent: Monday, October 19, 2009 3:46:07 PM<br>
Subject: RE: [VoiceOps] Shoretel 9.0 Call transfer using SIP
softphone Xlite/Eyebeam<br>
<br>
<br>
</span><span style='font-size:12.0pt;font-family:"Times New Roman","serif";
color:black'><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'> </span><span
style='color:black'><o:p></o:p></span></p>
<div>
<p class=MsoNormal><span style='color:#1F497D'>Here is what I see
happening…This is from ShoreGear Switch…</span><span style='color:black'><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>It seems like on the SIP Trunk I
am setup to prefix all calls with a 9, but when I transfer using Eyebeam
Shoretel messes up the number (converts 93035551000 to 9+13035551000) in the
REFER msg to the Broadsoft platform, hence call fails…</span><span
style='color:black'><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'> </span><span
style='color:black'><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>If I press Xfer on EBeam and
then dial 13035551000 (Don’t prefix it with a 9), it seems to work…</span><span
style='color:black'><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'> </span><span
style='color:black'><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>I think it’s a setting on
the Shoretel….Any ideas..?</span><span style='color:black'><o:p></o:p></span></p>
</div>
<p class=MsoNormal><span style='color:#1F497D'> </span><span
style='color:black'><o:p></o:p></span></p>
<div>
<div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0in 0in 0in;
border-color:-moz-use-text-color -moz-use-text-color'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:black'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:black'> Jason L. Nesheim [mailto:jnesheim@cytek.biz] <br>
<b>Sent:</b> Monday, October 19, 2009 4:31 PM<br>
<b>To:</b> Ujjval Karihaloo<br>
<b>Cc:</b> VoiceOps@voiceops.org<br>
<b>Subject:</b> Re: [VoiceOps] Shoretel 9.0 Call transfer using SIP softphone
Xlite/Eyebeam</span><span style='color:black'><o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><span style='color:black'> <o:p></o:p></span></p>
<div>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='color:black'>Could
you provide a SIP signaling capture of the communication between the Shoretel
PBX, Ingate, and Broadworks?<br>
<br>
There are several possible issues but it would be difficult to say what applies
to your scenario without a capture.<o:p></o:p></span></p>
<div>
<div>
<div>
<p class=MsoNormal><span style='color:black'>-- <br>
Jason Nesheim<o:p></o:p></span></p>
</div>
</div>
</div>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='color:black'><br>
<br>
----- Original Message -----<br>
From: "Ujjval Karihaloo" <ujjval@simplesignal.com><br>
To: VoiceOps@voiceops.org<br>
Sent: Monday, October 19, 2009 9:19:21 AM<br>
Subject: [VoiceOps] Shoretel 9.0 Call transfer using SIP
softphone Xlite/Eyebeam<br>
<br>
<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:black'>I am testing Shoretel shoretel
with Broadsoft and Call transfer seems to fail.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:black'> <o:p></o:p></span></p>
<p class=MsoNormal><span style='color:black'>Eyebeam</span><span
style='font-family:Wingdings;color:black'>à</span><span style='color:black'>
Shoregear</span><span style='font-family:Wingdings;color:black'>à</span><span
style='color:black'> SIP Provider(Broadsoft)</span><span style='font-family:
Wingdings;color:black'>à</span><span style='color:black'> PSTN<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:black'> <o:p></o:p></span></p>
<p class=MsoNormal><span style='color:black'>Eyebeam use calls a cell phone on the
PSTN side and then tried to transfer the call to PSTN user#2, Has anyone tested
this with a SIP Softphone registered to Shoretel ?<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:12.0pt;font-family:"Times New Roman","serif";
color:black'><br>
_______________________________________________<br>
VoiceOps mailing list<br>
VoiceOps@voiceops.org<br>
https://puck.nether.net/mailman/listinfo/voiceops</span><span style='color:
black'><o:p></o:p></span></p>
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