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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I would do a CCSIP debug. More likely than not, in this case,
what I’ve seen is a mismatch in a feature / command used for the SIP call
setup. Either a reinvite message is being misunderstood, or a new call should
be created versus the call getting forwarded. This would coincide with what
Mark is talking about with respect to call progress issues.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>MT<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<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"'>
cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>On
Behalf Of </b>Joel Perez<br>
<b>Sent:</b> Monday, January 25, 2010 11:13 AM<br>
<b>To:</b> Mark Holloway<br>
<b>Cc:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] Forwarded Calls drop after 29 secs<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<p class=MsoNormal>Hey Guys,<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>Thanks for the responses. The carrier still hasnt gotten
back to me yet with any results. <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>The carrier is using a Broadsoft platform so I will mention
this to them and have them take a look at it.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>Thanks, <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
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<p class=MsoNormal style='margin-bottom:12.0pt'>Joel P<o:p></o:p></p>
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<div>
<p class=MsoNormal>On Sun, Jan 24, 2010 at 12:10 PM, Mark Holloway <<a
href="mailto:mh@markholloway.com">mh@markholloway.com</a>> wrote:<o:p></o:p></p>
<div>
<p class=MsoNormal>The issue with 'call drops on forwarded calls' is usually a
result of the PBX treating the forwarded portion of the call as a totally
separate call leg from the originating call into the PBX and the PBX is not
providing a progress indicator to the telco switch for the original call leg
until the second call leg is answered. The telco switch will timeout if
it doesn't receive a progress message and drop the call because there has been
no acknowledgement to the original call setup. The reason it may not happen
100% of the time is if the forwarded call to the PSTN is setup or answered fast
enough, the PBX will notify the original call leg know the call has been
answered or there is ring back. Depending on the far end carrier your are
forwarding calls to, it may work in some instances but not others. Proper
PBX behavior is when calls are forwarded from the PBX to the PSTN, the PBX
should provide SIP Diversion or ISDN Progress Indicator on the original call
leg so the telco switch does not timeout. Carriers who are using Sonus,
Broadsoft, Metaswitch, often make static changes in their switch to work around
this problem, but the PBX is still doing it wrong. <o:p></o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<div>
<p class=MsoNormal>On Jan 23, 2010, at 3:20 PM, James Buchanan wrote:<o:p></o:p></p>
</div>
<p class=MsoNormal><br>
<br>
<o:p></o:p></p>
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<div>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I had the same issue once. The SIP provider’s provider was
killing the call at thirty seconds.</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-family:"Calibri","sans-serif";color:#1F497D'>James
Buchanan, CCIE #25863</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-family:"Calibri","sans-serif";color:#1F497D'>Senior
Network Engineer</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-family:"Calibri","sans-serif";color:#1F497D'>Coleman
Technologies, Inc.</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-family:"Calibri","sans-serif";color:#1F497D'>12
Cadillac Drive, Suite 130</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-family:"Calibri","sans-serif";color:#1F497D'>Brentwood,
TN 37017</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-family:"Calibri","sans-serif";color:#1F497D'>(615)
866-5729</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> </span><o:p></o:p></p>
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<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"'> <a
href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a
href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <b>On
Behalf Of </b>Jim Skelton<br>
<b>Sent:</b> Friday, January 22, 2010 2:46 PM<br>
<b>To:</b> Joel Perez<br>
<b>Cc:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> Re: [cisco-voip] Forwarded Calls drop after 29 secs</span><o:p></o:p></p>
</div>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<div>
<p class=MsoNormal>Hi Joel, I had a similar issue. I would see our
calls drop after exactly 30 seconds. It turned out to be an issue with
our carrier TimeWarner telecom and thier Oakland SIP switch. We
repointed to thier North Carolina SIP switch and everything cleared
up. TW is running Sonus switches.<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal> Our setup with twtelecom is
Carrier---sip---CUBE----h323---CUCM<o:p></o:p></p>
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<div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
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<div>
<div>
<p class=MsoNormal>Hope this helps.<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>Jim Skelton<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>Halliburton<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal><br>
<br>
<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>On Fri, Jan 22, 2010 at 1:13 PM, Joel Perez <<a
href="mailto:tman701@gmail.com" target="_blank">tman701@gmail.com</a>>
wrote:<o:p></o:p></p>
</div>
<div>
<div>
<p class=MsoNormal>Hey Guys,<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>I have a customer that is experiencing a weird issue.<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>Currently some of their forwarded calls are failing after 29
secs. I believe it is only happening when the calls are forwarded to 2 specific
carriers, but cant confirm that yet.<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>I have a ticket open with the carrier so that they can take
a look at the traces we have captured.<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>The issue happens the following way. Inbound call to main
company DID goes to an AA, if no choice is used by the caller then it goes to a
live person. However afterhours this live person forwards their call to an
offsite #. When that offsite # receives the forwarded call they are only able
to stay on for 29 secs then the call dies on both ends.<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>THe set up is as follows:<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>Carrier---sip---CUBE----sip---CUCM---sccp---IPT<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>CUCM is 7.0<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>Unity is 7.0<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>CUBE is 12.4.(20T4)<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>I have tried capturing debugs on the CUBE but havent been
able to see any SIP (BYE) messages from either side.<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>This only happens when Unity is involved. Normal CFW doesnt
have this problem.<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>Thanks,<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=MsoNormal>Joel P<o:p></o:p></p>
</div>
</div>
<p class=MsoNormal style='margin-bottom:12.0pt'><br>
_______________________________________________<br>
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<a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><br>
<br clear=all>
<br>
-- <br>
“A government big enough to give you everything you want, is<br>
strong enough to take everything you have.” — Thomas Jefferson<o:p></o:p></p>
</div>
</div>
<p class=MsoNormal><span style='font-size:13.5pt;font-family:"Helvetica","sans-serif"'>_______________________________________________<br>
cisco-voip mailing list<br>
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<a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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