Jeff,<br><br>I'd do what Wes is saying, however I will check the traces myself before I send those to TAC.<br>Trace the call look for the StationOpenReceiveChannel output, this contains the media Payload and bit rate check the timestamp of this message, then look for this message StationOpenReceiveChannelAck and then look for StartMediaTransmission, this message commands the phone to start streaming RTPs, this includes UDP port, IP of remote endpoint, packet size and codec. check the timestamps and if the delay between those messages is greater than 40ms or 200ms I don't remember hopefully someone knows the roundtrip delay for MGCP. then the problem is MGCP and you may want to consider H.323.<br>
<br>I have an entire call flow from a few tests calls I made and got all the messages that are involved<br><br><meta http-equiv="Content-Type" content="text/html; charset=utf-8"><meta name="ProgId" content="OneNote.File"><meta name="Generator" content="Microsoft OneNote 12">
<p style="margin: 0in; font-weight: bold; font-family: Calibri; font-size: 11pt;">Call
Processing Behavior</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;"> </p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">P->CM:<span style=""> </span>OffHookMessage</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationCallState:<span style=""> </span>OffHook (1)</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:<span style=""> </span>StationDisplayPromptStatus:<span style=""> </span>update the display </p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationSelectSoftKeys: loads the appropiate soft key set, depending on the call
state</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationActivateCallPlane:<span style=""> </span>contains the
specified line appearance of the DN being called</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationStartTone: InsideDialTone,<span style="">
</span>commands the phone to play dialtone</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">P->CM:
KeyPadButton: keypad button was pressed. *= 0xe; #=0xf</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:<span style=""> </span>StationStopTone: msg sent when a tone needs
to be stopped (i.e. DialTone)</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationCallState:<span style=""> </span>State of the call:
OffHook=1, OnHook=2, RingOut=3, RingIn=4, Connected=5</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;"><span style=""> </span>Busy=6, Congestion=7,
Hold=8, CallWaiting=9, CallTransfer=10, CallPark=11, Proceed=12</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;"><span style=""> </span>CallRemoteMultiline=13,
InvalidNumber=14</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:<span style=""> </span>StationStartTone:<span style=""> </span>(outsidedialtone=34)</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">P->CM:<span style=""> </span>keyPadButton:</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:<span style=""> </span>StationStopTone</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:<span style=""> </span>StationCallInfo:<span style=""> </span>msg has the called party DN/Name and Calling
party DN/Name</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationSetRinger:<span style=""> </span>sets the ringer to the
specified ringing mode: StationRingOff: stops ringer from Ringing,
StationInsideRinging: indicates OnNetCall,</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;"><span style=""> </span>StationOutsideRing:
indicates OffNetCall, StationFeatureRing: used by third-party apps to invoke
special features</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationDisplayNotify:<span style=""> </span>this msg causes
the phone to discard msg txt from StationOutputDispalyText and play the text
contained in StationDisplayNotify</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationDisplayPromptStatus</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationSelectSoftKeys</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationCallState:<span style=""> </span>proceed=12</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationCallInfo</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationStartTone: alerting Tone,<span style="">
</span>ringback tone</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationCallState: ringout=3</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationSelectSoftKeys</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationDisplayPromptStatus</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">P2->CM:
OffHookMessage</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationClearNotify: msg sent to the phone to clear the information sent in the
StationDisplayNotify msg</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationSetRinger: RingerOff</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationCallState</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationActivateCallPlane</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationStopTone</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationCallState</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationCallInfo</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationSelectSoftkeys</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationDisplayPromptStatus</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationOpenReceiveChannel: contains the media Payload and bit rate, asks the
phone if it is ready to receive RTP Stream</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationOpenReceiveChannel</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">P->CM:
StationOpenReceiveChannelAck</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">P2->CM:
StationOpenReceiveChannelAck</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StartMediaTransmission: commands the phone to start streaming RTP. Includes:
UDP port , IP of remote endpoint, packet size, Codec</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StartMediaTransmission</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">P2->CM:
OnHookMessage</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationConnectionStatisticsReq:<span style=""> </span>requests
connectionstatistics from the ip phone</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">P2->CM:
StationSetSpeakerMode: turns the speakerphone on/off</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationClearStatus: </p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationCallState: 2=onhook</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationSelectSoftkeys</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationDisplayPromptStatus</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationActivateCallPlane</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">P2->CM:
StationConnectionStatisticsRes</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationDefineTimeDate</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationStopTone</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationCloseReceiveChannel:<span style=""> </span>commands the
phone to stop processing RTP messages sent to it</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P:
StationStopMediaTransmission: tells the phone to stop streaming RTP packets </p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationCloseReceiveChannel</p>
<p style="margin: 0in; font-family: Calibri; font-size: 11pt;">CM->P2:
StationStopMediaTransmission</p>
<br><br>CM is The communications Manager<br>P is phone 1 the phone that is placing the call<br>P2 is the phone that will receive the call<br><br>hopefully this will help you out<br><br><br>Cris<br><br><div class="gmail_quote">
2009/6/25 Wes Sisk <span dir="ltr"><<a href="mailto:wsisk@cisco.com">wsisk@cisco.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000">
Unfortunately that is not quite the whole picture. DNA does not
gracefully identify and handle all overlap conditions which can cause
delayed routing.<br>
<br>
Overlap dial plan is most likely. However, there are other signaling
issues such as long round trip time that can caused delayed call
routing. Another cause is hunting through endpoints in a routelist or
routegroup which are partially responsive.<br>
<br>
CCM SDI and SDL traces from the involved CM servers is the best way to
identify what is introducing the delay. If you are uncomfortable
reviewing those open a TAC case and attach them. Include on the case:<br>
traces<br>
calling party number<br>
called party number<br>
approximate time of call based on phone time.<br>
phone time offset from the server<br><font color="#888888">
<br>
<br>
/Wes</font><div><div></div><div class="h5"><br>
<br>
<br>
On Thursday, June 25, 2009 2:16:09 PM, Jeff Ruttman
<a href="mailto:ruttmanj@carewisc.org" target="_blank"><ruttmanj@carewisc.org></a> wrote:<br>
</div></div><blockquote type="cite"><div><div></div><div class="h5">
<div dir="ltr" align="left"><font size="2" color="#0000ff" face="Arial"><span>Thanks Cristobal, Ian.</span></font></div>
<div dir="ltr" align="left"><font size="2" color="#0000ff" face="Arial"><span></span></font> </div>
<div dir="ltr" align="left"><font size="2" color="#0000ff" face="Arial"><span>There doesn't appear to be a
dial plan problem. See below.</span></font></div>
<div dir="ltr" align="left"><font size="2" color="#0000ff" face="Arial"><span></span></font> </div>
<div dir="ltr" align="left"><font size="2" color="#0000ff" face="Arial"><span>I'm open to changing
protocol to H.323, but I wouldn't know how to do that exactly at the
moment. And anyway as I mentioned we have an H.323 GW for each of the
sites that use Trunks and POTS. They've always had a status of
"unknown" and if you view their config, the registration is unknown. 2
of the 4 have a device name/IP address that is the same as the routers
at those sites, and the other 2 have a device name/IP address that is
the same as the IP on the FXO and FXS CCM configs on the MGCP GWs for
these sites. They are otherwise configured the same.</span></font></div>
<div dir="ltr" align="left"><font size="2" color="#0000ff" face="Arial"><span></span></font> </div>
<div dir="ltr" align="left"><font size="2" color="#0000ff" face="Arial"><span>Are these H.323 GWs just
ornamental?? Are they doing anything? Certainly the MGCP GWs are....</span></font></div>
<div dir="ltr" align="left"><font size="2" color="#0000ff" face="Arial"><span></span></font> </div>
<div dir="ltr" align="left"><font size="2" color="#0000ff" face="Arial"><span>I'll keep plugging away.
Little by little I'll catch on.</span></font></div>
<div dir="ltr" align="left"><font size="2" color="#0000ff" face="Arial"><span></span></font> </div>
<div dir="ltr" align="left"><font size="2" color="#0000ff" face="Arial"><span>Thanks</span></font></div>
<div dir="ltr" align="left"><font size="2" color="#0000ff" face="Arial"><span>jeff</span></font></div>
<div dir="ltr" align="left"><font size="2" color="#0000ff" face="Arial"><span></span></font> </div>
<div dir="ltr" align="left"><font size="2" color="#0000ff" face="Arial"><span>
<li><b title="Results Summary">Results Summary</b>
<ul>
<li><b title="Calling Party Information">Calling
Party Information</b>
<ul>
<li><b title="Calling Party">Calling Party</b> =
5801 <br>
</li>
<li><b title="Partition">Partition</b> = <br>
</li>
<li><b title="Device CSS">Device CSS</b> = <br>
</li>
<li><b title="Line CSS">Line CSS</b> = MO1Phones <br>
</li>
<li><b title="AAR Group Name">AAR Group Name</b> = <br>
</li>
<li><b title="AAR Calling Search Space">AAR CSS</b>
= <br>
</li>
</ul>
</li>
<li><b title="Dialed Digits">Dialed Digits</b> =
92977902 </li>
<li><b title="Match Result">Match Result</b> =
RouteThisPattern </li>
<li><b title="Matched Pattern Information">Matched
Pattern Information</b>
<ul>
<li><b title="Pattern">Pattern</b> = 9.[2-9]XXXXXX <br>
</li>
<li><b title="Partition">Partition</b> = MO1Routes <br>
</li>
<li><b title="Time Schedule">Time Schedule</b> = <br>
</li>
</ul>
</li>
<li><b title="Called Party Number">Called Party Number</b>
= 92977902 </li>
<li><b title="Time Zone">Time Zone</b> = Central
Standard/Daylight Time </li>
<li><b title="End Device">End Device</b> = MO1-RL-Local
</li>
<li><b title="Device Classification">Call Classification</b>
= OffNet </li>
<li><b title="InterDigit Timeout">InterDigit Timeout</b>
= NO </li>
<li><b title="Device Override">Device Override</b> =
Disabled </li>
<li><b title="Outside Dial Tone">Outside Dial Tone</b>
= NO </li>
</ul>
</li>
</span></font></div>
<br>
<div dir="ltr" align="left" lang="en-us">
<hr><font size="2" face="Tahoma"><b>From:</b> Cristobal
Priego [<a href="mailto:cristobalpriego@gmail.com" target="_blank">mailto:cristobalpriego@gmail.com</a>] <br>
<b>Sent:</b> Thursday, June 25, 2009 11:28 AM<br>
<b>To:</b> Ian MacKinnon<br>
<b>Cc:</b> Jeff Ruttman; cisco-voip<br>
<b>Subject:</b> Re: [cisco-voip] Slow to connect calls<br>
</font><br>
</div>
Sounds like Ian is right. you can have a dial plan problem<br>
do you have a centralized deployment? if you do, you need to be very
careful with the delay of MGCP, because this is a Master-Slave
protocol. MGCP has a dependency of callmanager, so before you can place
a call, the gw needs to talk to the CUCM to know what to do. if that's
the case I'd say the best option for you is to change your protocol to
H.323.<br>
<br>
<br>
Cris<br>
<br>
<div class="gmail_quote">2009/6/25 Ian MacKinnon <span dir="ltr"><<a href="mailto:Ian.Mackinnon@lumison.net" target="_blank">Ian.Mackinnon@lumison.net</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div vlink="purple" link="blue" lang="EN-GB">
<div>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Hi Jeff,</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">That
sounds like a dial plan problem ie it is waiting for another digit, and
then timing out.</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"></span> </p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Can you
dial the number before hitting dial on the phone so it is all present
as opposed to lifting the handset and dialling each digit in turn?</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"></span> </p>
<div>
<div style="border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0cm 0cm;">
<p><b><span style="font-size: 10pt;" lang="EN-US">From:</span></b><span style="font-size: 10pt;" lang="EN-US"> <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>Jeff Ruttman<br>
<b>Sent:</b> 25 June 2009 14:47<br>
<b>To:</b> cisco-voip<br>
<b>Subject:</b> [cisco-voip] Slow to connect calls</span></p>
</div>
</div>
<div>
<div>
<p> </p>
<div>
<p><span style="font-size: 10pt;">Greetings,</span></p>
</div>
<div>
<p> </p>
</div>
<div>
<p><span style="font-size: 10pt;">Some of our sites have DID trunk
ports and POTS lines, and we have MGCP controlled GWs with FXS and FXO
configured. We also have for these sites H.323 GWs--which frankly I'm
not sure why or what they do.</span></p>
</div>
<div>
<p> </p>
</div>
<div>
<p><span style="font-size: 10pt;">Anyway, at one of those sites, it
takes a count of 15 or more for an outgoing call to connect. I know
some delay is expected with that setup, but that's quite a bit longer
than at our comparable sites.</span></p>
</div>
<div>
<p> </p>
</div>
<div>
<p><span style="font-size: 10pt;">Is that length of delay still
within expectations? Or is there something perhaps I can do to speed
that up?</span></p>
</div>
<div>
<p> </p>
</div>
<div>
<p><span style="font-size: 10pt;">Thanks</span></p>
</div>
<div>
<p><span style="font-size: 10pt;">jeff</span></p>
</div>
<div>
<p> </p>
</div>
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