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<DIV dir=ltr align=left><SPAN class=484364301-27012006><FONT face=Arial
color=#0000ff size=2>I noticed that the normal dial-peers with "application
MGCPAPP" disappeared I think when I first put 12.3 on a router. Is it
possible MGCP with config-download now tucks itself on the default
peer?</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] <B>On Behalf Of </B>Kevin
Thorngren<BR><B>Sent:</B> Thursday, January 26, 2006 7:03 PM<BR><B>To:</B>
James Grace<BR><B>Cc:</B> CiscosupportUpuck
<cisco-voip@puck.nether.net><BR><B>Subject:</B> Re: [cisco-voip]
transcoding on srst gw<BR></FONT><BR></DIV>
<DIV></DIV>I responded too quickly, too late in the evening :-) The default
dial peer is matched for inbound calls. Try using "debug voip ccapi inout" to
see what is happening with the call. Maybe we can take a look at this along
with your config (sanitize as needed).<BR><BR>Kevin<BR>On Jan 26, 2006, at
7:53 PM, Kevin Thorngren wrote:<BR><BR>
<BLOCKQUOTE>You are matching the default dial peer which is hidden. Please
read the
following:<BR>http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml<BR><BR>Kevin<BR>On
Jan 26, 2005, at 5:57 PM, James Grace wrote:<BR><BR>
<BLOCKQUOTE><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>Ok
well the concern is that all call are working fine. Even thou
we do not see the dialpeer. So we are clueless on how calls are
going out with out the dial peer. We know that they are going out
the right gw but doing debugs q931 921<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>From:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
Voll, Scott
[mailto:Scott.Voll@wesd.org]<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Sent:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
Thursday, January 26, 2006 5:21 PM<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>To:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
James Grace<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Subject:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
RE: [cisco-voip] transcoding on srst gw<?/x-tad-bigger><?/fontfamily><BR><?fontfamily><?param Times New Roman><?bigger><?bigger> <?/bigger><?/bigger><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>I
think your other engineer is right. But I don’t have any doc’s to
back it up.<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>I
know on my 3640 with PRI I put in a pots dial peer of 9991024 as my
PRI is in slot1 port 0.<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>dial-peer
voice 9991024 pots<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> destination-pattern
9T<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> incoming
called-number .<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> direct-inward-dial<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>I
do remember manually putting it in. <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>Scott<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>Sorry,
it’s been years.<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>From:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
James Grace [mailto:jgrace@digitelusa.net]
<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Sent:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
Wednesday, January 26, 2005 10:48 AM<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>To:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
Voll, Scott<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Subject:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
RE: [cisco-voip] transcoding on srst gw<?/x-tad-bigger><?/fontfamily><BR><?fontfamily><?param Times New Roman><?bigger><?bigger> <?/bigger><?/bigger><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>Hey
Voll you seem to have all the answers. So I thank you for all the
help.. I have a question that me and another engineer have a
issue with. Can you explain to me or help me understand
why on a mgcp gw, we do not see a dial peer in the
config for the pri endpoint ( like you would see after confuring fxo and
fxs end points in cm) and how would I explain why there isn’t
suppose to be one in the cli conf. my other engineer seems to thing
that you should add one manully to the cli gw<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>From:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
Voll, Scott
[mailto:Scott.Voll@wesd.org]<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Sent:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
Thursday, January 26, 2006 12:19 PM<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>To:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
James Grace<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Subject:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
RE: [cisco-voip] transcoding on srst gw<?/x-tad-bigger><?/fontfamily><BR><?fontfamily><?param Times New Roman><?bigger><?bigger> <?/bigger><?/bigger><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>But
is the conference calling resource at the remote site or the central
site. If it’s at the central site then the transcoding is just need
at the central site. Now if the conference calling resource is at
the remote site, yes you will need transcoding at the remote site.<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>Scott<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>From:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
James Grace [mailto:jgrace@digitelusa.net]
<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Sent:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
Wednesday, January 26, 2005 8:56 AM<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>To:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
Voll, Scott<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Subject:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
RE: [cisco-voip] transcoding on srst gw<?/x-tad-bigger><?/fontfamily><BR><?fontfamily><?param Times New Roman><?bigger><?bigger> <?/bigger><?/bigger><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>Yes
we are doing conf call internal at the srst and conf calling between the
srst and main. <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>From:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
Voll, Scott
[mailto:Scott.Voll@wesd.org]<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Sent:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
Thursday, January 26, 2006 11:36 AM<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>To:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
James Grace<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Subject:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
RE: [cisco-voip] transcoding on srst gw<?/x-tad-bigger><?/fontfamily><BR><?fontfamily><?param Times New Roman><?bigger><?bigger> <?/bigger><?/bigger><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>James—<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>You
are going to setup regions to your remote sites so you can use G729.
That I understand. Do you have some kind of resource at the remote
site that only does G711 that you’re other sites need to get too?
That would be the only reason I can think of for needing transcoding
resources at the remote
site. <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>I
have transcoding resources at the central site because I have IPCCx that
only does G711. So if a remote site is using G729 the central site
transcodes it to G711 so they can make the call to the IPCCx pilot point.<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>Are
you doing conferencing calling resources at the remote site that would
require transcoding at the remote site?<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>Scott<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>From:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
James Grace [mailto:jgrace@digitelusa.net]
<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Sent:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
Wednesday, January 26, 2005 8:25 AM<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>To:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
Voll, Scott<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Subject:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
RE: [cisco-voip] transcoding on srst gw<?/x-tad-bigger><?/fontfamily><BR><?fontfamily><?param Times New Roman><?bigger><?bigger> <?/bigger><?/bigger><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>First
thanks for the reply<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>I
was told that i need to set up region for srst with codec 7.29 and set up
transcoding on the srst gw to dedicate a few dsp fo r transcoding<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>From:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
Voll, Scott
[mailto:Scott.Voll@wesd.org]<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Sent:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
Thursday, January 26, 2006 11:03 AM<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>To:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
James Grace;
cisco-voip@puck.nether.net<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Subject:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
RE: [cisco-voip] transcoding on srst gw<?/x-tad-bigger><?/fontfamily><BR><?fontfamily><?param Times New Roman><?bigger><?bigger> <?/bigger><?/bigger><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>What
is the reason for having transcoding on your SRST gw?<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>If
your in SRST you would be using the PSTN. <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger>Scott<?/x-tad-bigger><?/color><?/fontfamily><BR><?fontfamily><?param Arial><?color><?param 0000,0000,807F><?x-tad-bigger> <?/x-tad-bigger><?/color><?/fontfamily><BR><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>From:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] <?/x-tad-bigger><B><?x-tad-bigger>On Behalf Of <?/x-tad-bigger></B><?x-tad-bigger>James Grace<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Sent:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
Wednesday, January 26, 2005 7:37 AM<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>To:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
cisco-voip@puck.nether.net<?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Tahoma><?x-tad-bigger>Subject:<?/x-tad-bigger><?/fontfamily></B><?fontfamily><?param Tahoma><?x-tad-bigger>
[cisco-voip] transcoding on srst gw<?/x-tad-bigger><?/fontfamily><BR><?fontfamily><?param Times New Roman><?bigger><?bigger> <?/bigger><?/bigger><?/fontfamily><BR><?fontfamily><?param Arial><?x-tad-bigger>I
was told that I need to configure transcoding on my srst gw. And not
sure way. Is this really needed. And if I set up a media
resource groups for my srst site, the available resources are on the cms
at the main site. So the question is if I create one for srst and
use the avil resource isn’t that the same as using none or the default<?/x-tad-bigger><?/fontfamily><BR><?fontfamily><?param Arial><?x-tad-bigger> <?/x-tad-bigger><?/fontfamily><BR><?fontfamily><?param Arial><?bigger><?bigger><?bigger>James
D. Grace<?/bigger><?/bigger><?/bigger><?x-tad-bigger> <?/x-tad-bigger><?/fontfamily><BR><B><?fontfamily><?param Arial><?x-tad-bigger>CCNP
CCNA MCSE MCDBA<?/x-tad-bigger><?/fontfamily></B><BR><?fontfamily><?param Arial><?x-tad-bigger>Sr.
System Engineer / Professional Svc.<?/x-tad-bigger><?/fontfamily><BR><B><?bigger><?bigger>Digitel
Corporation<?/bigger><?/bigger></B><BR><?fontfamily><?param Times New Roman><?bigger><?bigger> <?/bigger><?/bigger><?/fontfamily><BR>_______________________________________________<BR>cisco-voip
mailing
list<BR>cisco-voip@puck.nether.net<BR>https://puck.nether.net/mailman/listinfo/cisco-voip<BR></BLOCKQUOTE>_______________________________________________<BR>cisco-voip
mailing
list<BR>cisco-voip@puck.nether.net<BR>https://puck.nether.net/mailman/listinfo/cisco-voip<BR></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>