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Hi Adam,<br>
<br>
As a courtesy to everyone in the community please do not hijack
existing threads. For a new question please begin a new thread. I
have not seen or heard any limit placed on number of active
discussions. If you are encountering any limitation please let us know
so we can take it up with the moderators and admins.<br>
<br>
Regards,<br>
Wes<br>
<br>
On Friday, July 31, 2009 10:04:38 AM , Curtis, Adam
<a class="moz-txt-link-rfc2396E" href="mailto:ACurtis@pizzagalli.com"><ACurtis@pizzagalli.com></a> wrote:<br>
<blockquote
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<div class="Section1">
<p class="MsoNormal"><span style="color: rgb(31, 73, 125);">Over the
couple of days our
users have noticed that if they are on a call and want to conference
someone
else in they press the <b>Confrn</b> button, dial the number and the
person
answers, but when they press the <b>Confrn</b> button again one or
both parties
are dropped from the call. Is this a codec issue? If so, what log
traces should
I be looking for? I tried this with a remote site, cell phone, and
internal
phones and it always disconnects but if I conference in voicemail and
they auto
attendant it works fine.. Any ideas would be greatly appreciated!<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color: rgb(31, 73, 125);"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color: rgb(31, 73, 125);">Thank you,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color: rgb(31, 73, 125);"><o:p> </o:p></span></p>
<p class="MsoNormal"><b><span
style="font-size: 10pt; color: rgb(31, 73, 125);">Adam Curtis<o:p></o:p></span></b></p>
<p class="MsoNormal"><b><span
style="font-size: 10pt; color: rgb(31, 73, 125);">Pizzagalli
Construction Company<o:p></o:p></span></b></p>
<p class="MsoNormal"><b><span
style="font-size: 10pt; color: rgb(31, 73, 125);">IT Support
Technician<o:p></o:p></span></b></p>
<p class="MsoNormal"><b><span
style="font-size: 10pt; color: rgb(31, 73, 125);">(802)
651-1319<o:p></o:p></span></b></p>
<p class="MsoNormal"><span style="color: rgb(31, 73, 125);"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color: rgb(31, 73, 125);"><o:p> </o:p></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 0in 0in;">
<p class="MsoNormal"><b><span
style="font-size: 10pt; font-family: "Tahoma","sans-serif"; color: windowtext;">From:</span></b><span
style="font-size: 10pt; font-family: "Tahoma","sans-serif"; color: windowtext;">
<a class="moz-txt-link-abbreviated" href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>
[<a class="moz-txt-link-freetext" href="mailto:cisco-voip-bounces@puck.nether.net">mailto:cisco-voip-bounces@puck.nether.net</a>] <b>On Behalf Of </b>Wes
Sisk<br>
<b>Sent:</b> Thursday, July 30, 2009 10:31 AM<br>
<b>To:</b> Leslie Meade<br>
<b>Cc:</b> <a class="moz-txt-link-abbreviated" href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> Re: [cisco-voip] intermittent missing outbound ring
tone<o:p></o:p></span></p>
</div>
</div>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Hi Leslie,<br>
<br>
The next messages after setup are the ones that determine how the user
hears
ringback. If the PSTN sends Proceeding, Alerting, or Facility with a
Progress Indicator IE and that Progress Indicator ends in 8 then the
IOS
gateway and CM must cut through audio immediately. This allows the
user
to hear any inband tones provided by the transit PSTN gear or by the
far end
customer premise equipment.<br>
<br>
I am not aware of a single one-size-fits-all solution to this. Some
services, such as American Express, send PI=8 to play prompts to caller
at the
beginning of the call. In this case it is necessary to cut through the
audio. You may call other CPE equipment that provide PI=8 but then
fail
to provide inband audio of either announcement or ringback.<br>
<br>
That said there are a few tools at your disposal:<br>
<a moz-do-not-send="true"
href="http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a0080094c33.shtml">http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a0080094c33.shtml</a><br>
<br>
/Wes<br>
<br>
On Wednesday, July 29, 2009 6:07:07 PM , Leslie Meade <a
moz-do-not-send="true" href="mailto:lmeade@signal.ca"><lmeade@signal.ca></a>
wrote:<br>
<br>
<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Since converting my gateways to H.323, I am
getting users
reporting that sometimes they do not get a ring tone when they ring
people.<o:p></o:p></p>
<p class="MsoNormal">It is not every time, not every number. It can
be long
distance or international, it does not happen all the time. Any ideas<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">I am running ccm7.1 and my gateway is a 2821
H.323<o:p></o:p></p>
<p class="MsoNormal"><br>
4743212: *Jul 29 14:42:22.891 UTC: ISDN Se0/0/0:23 Q931:
pak_private_number:
Invalid type/plan 0x0 0x0 may be overriden; sw-type 13<o:p></o:p></p>
<p class="MsoNormal">4743213: *Jul 29 14:42:22.891 UTC: ISDN
Se0/0/0:23 Q931:
pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type
13<o:p></o:p></p>
<p class="MsoNormal">4743214: *Jul 29 14:42:22.891 UTC: ISDN
Se0/0/0:23 Q931:
Sending SETUP callref = 0x5AF1 callID = 0xDBB7 switch = primary-ni
interface = User <o:p></o:p></p>
<p class="MsoNormal">4743215: *Jul 29 14:42:22.895 UTC: ISDN
Se0/0/0:23 Q931: TX
-> SETUP pd = 8 callref = 0x5AF1 <o:p></o:p></p>
<p class="MsoNormal"> Bearer
Capability i = 0x8090A2 <o:p></o:p></p>
<p class="MsoNormal"> Standard
= CCITT <o:p></o:p></p>
<p class="MsoNormal"> Transfer
Capability = Speech <o:p></o:p></p>
<p class="MsoNormal"> Transfer
Mode = Circuit <o:p></o:p></p>
<p class="MsoNormal"> Transfer
Rate = 64 kbit/s <o:p></o:p></p>
<p class="MsoNormal"> Channel ID i
= 0xA9838B <o:p></o:p></p>
<p class="MsoNormal"> Exclusive,
Channel 11 <o:p></o:p></p>
<p class="MsoNormal"> Facility i =
0x9F8B0100A11502012E020100800D4B72797374792057696465656E <o:p></o:p></p>
<p class="MsoNormal"> Protocol
Profile = Networking Extensions <o:p></o:p></p>
<p class="MsoNormal"> 0xA11502012E020100800D4B72797374792057696465656E
<o:p></o:p></p>
<p class="MsoNormal"> Component
= Invoke component <o:p></o:p></p>
<p class="MsoNormal"> Invoke
Id = 46 <o:p></o:p></p>
<p class="MsoNormal"> Operation
= CallingName <o:p></o:p></p>
<p class="MsoNormal"> Name
Presentation Allowed Extended<o:p></o:p></p>
<p class="MsoNormal">
Name = Krysty <o:p></o:p></p>
<p class="MsoNormal"> Display i =
'Krysty’ <o:p></o:p></p>
<p class="MsoNormal"> Calling
Party Number i = 0x2181, '604899XXXX' <o:p></o:p></p>
<p class="MsoNormal"> Plan:ISDN,
Type:National <o:p></o:p></p>
<p class="MsoNormal"> Called Party
Number i = 0x80, '01152818421XXXX' <o:p></o:p></p>
<p class="MsoNormal"> Plan:Unknown,
Type:Unknown<o:p></o:p></p>
<p class="MsoNormal">4743216: *Jul 29 14:42:22.931 UTC: ISDN
Se0/0/0:23 Q931: RX
<- STATUS pd = 8 callref = 0xDAF1 <o:p></o:p></p>
<p class="MsoNormal"> Cause i =
0x80E328 - Information element not implemented<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Leslie<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<pre><o:p> </o:p></pre>
<pre style="text-align: center;">
<hr align="center" size="4" width="90%">
</pre>
<pre><o:p> </o:p></pre>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>cisco-voip mailing list<o:p></o:p></pre>
<pre><a moz-do-not-send="true"
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><o:p></o:p></pre>
<pre><a moz-do-not-send="true"
href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></pre>
<pre> <o:p></o:p></pre>
<p class="MsoNormal"><span
style="font-size: 12pt; font-family: "Times New Roman","serif";"><o:p> </o:p></span></p>
</div>
<pre wrap="">
<hr size="4" width="90%">
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