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<DIV dir=ltr align=left><SPAN class=401410818-06082008><FONT face=Arial
color=#0000ff size=2>Jonathan,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=401410818-06082008><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN
class=401410818-06082008> <FONT face=Arial color=#0000ff
size=2>I thinkyou need the command</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=401410818-06082008></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN
class=401410818-06082008> <FONT face=Arial color=#0000ff
size=2>redirect ip2ip under voice service voip (Global)
or under each Dial¨-Peer (specific)</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=401410818-06082008></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=401410818-06082008><FONT face=Arial
color=#0000ff size=2>Regards,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=401410818-06082008></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN
class=401410818-06082008> <FONT face=Arial color=#0000ff
size=2>Ariel</FONT> </SPAN></DIV><BR>
<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>Jonathan
Charles<BR><B>Sent:</B> Miércoles, 06 de Agosto de 2008 02:56 p.m.<BR><B>To:</B>
Nguyen Le<BR><B>Cc:</B> OSL CCIE Voice Lab Exam;
cisco-voip@puck.nether.net<BR><B>Subject:</B> Re: [cisco-voip] [OSL |
CCIE_Voice] IPIPGW Sip to Sip<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV dir=ltr>Call is from x1008 (on CCM) to x3003 (on CCME)<BR><BR>On the
IPIPGW:<BR><BR><BR>voice service voip <BR> allow-connections h323 to
h323<BR> allow-connections h323 to sip<BR> allow-connections sip to
h323<BR> allow-connections sip to sip<BR> h323<BR> modem
passthrough nse codec g711ulaw<BR> sip<BR><BR>dial-peer voice 500
voip<BR> destination-pattern 1008<BR> session protocol
sipv2<BR> session target ipv4:<A
href="http://10.0.0.124">10.0.0.124</A><BR> dtmf-relay sip-notify
rtp-nte<BR> codec g711ulaw<BR>!<BR>dial-peer voice 501
voip<BR> destination-pattern 3...<BR> session target ipv4:<A
href="http://10.0.0.131">10.0.0.131</A><BR> incoming called-number
3003<BR> codec g711ulaw<BR>!<BR><BR>On CCME:<BR><BR><BR>dial-peer voice 201
voip<BR> destination-pattern 1008<BR> session protocol
sipv2<BR> session target ipv4:<A
href="http://10.0.0.63">10.0.0.63</A><BR> incoming called-number
3003<BR> dtmf-relay rtp-nte<BR> codec
g711ulaw<BR><BR><BR><BR><BR>Jonathan<BR><BR>
<DIV class=gmail_quote>On Wed, Aug 6, 2008 at 12:54 PM, Nguyen Le <SPAN
dir=ltr><<A
href="mailto:nguyenbot@gmail.com">nguyenbot@gmail.com</A>></SPAN> wrote:<BR>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<DIV lang=EN-US vlink="purple" link="blue">
<DIV>
<P><SPAN style="FONT-SIZE: 11pt; COLOR: rgb(31,73,125)">You also have
under</SPAN></P>
<P><SPAN style="FONT-SIZE: 11pt; COLOR: rgb(31,73,125)">Voice service
voip</SPAN></P>
<P><SPAN style="FONT-SIZE: 11pt; COLOR: rgb(31,73,125)">Allow connections sip
to sip ?</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)">Also, just double
check and make sure your SIP Trunk is in a region that is set to G711 to all
other sites</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)">Nguyen</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)"></SPAN> </P>
<DIV
style="BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: rgb(181,196,223) 1pt solid; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: medium none; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<P><B><SPAN style="FONT-SIZE: 10pt">From:</SPAN></B><SPAN
style="FONT-SIZE: 10pt"> Jonathan Charles [mailto:<A
href="mailto:jonvoip@gmail.com" target=_blank>jonvoip@gmail.com</A>]
<BR><B>Sent:</B> Wednesday, August 06, 2008 12:52 PM<BR><B>To:</B> Nguyen
Le<BR><B>Cc:</B> OSL CCIE Voice Lab Exam; <A
href="mailto:cisco-voip@puck.nether.net"
target=_blank>cisco-voip@puck.nether.net</A><BR><B>Subject:</B> Re: [OSL |
CCIE_Voice] IPIPGW Sip to Sip</SPAN></P></DIV>
<DIV>
<DIV></DIV>
<DIV class=Wj3C7c>
<P> </P>
<DIV>
<P style="MARGIN-BOTTOM: 12pt">Yeah, all dial-peers have the codec hard set to
711ulaw<BR><BR><BR>Jonathan</P>
<DIV>
<P>On Wed, Aug 6, 2008 at 12:43 PM, Nguyen Le <<A
href="mailto:nguyenbot@gmail.com" target=_blank>nguyenbot@gmail.com</A>>
wrote:</P>
<DIV>
<DIV>
<P><SPAN style="FONT-SIZE: 11pt; COLOR: rgb(31,73,125)">Jonathan – </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)">Make sure your call
codec is g711</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)">Nguyen</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)"></SPAN> </P>
<DIV
style="BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: 1pt solid; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: medium none; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<P><B><SPAN style="FONT-SIZE: 10pt">From:</SPAN></B><SPAN
style="FONT-SIZE: 10pt"> <A
href="mailto:ccie_voice-bounces@onlinestudylist.com"
target=_blank>ccie_voice-bounces@onlinestudylist.com</A> [mailto:<A
href="mailto:ccie_voice-bounces@onlinestudylist.com"
target=_blank>ccie_voice-bounces@onlinestudylist.com</A>] <B>On Behalf Of
</B>Jonathan Charles<BR><B>Sent:</B> Wednesday, August 06, 2008 12:41
PM<BR><B>To:</B> OSL CCIE Voice Lab Exam; <A
href="mailto:cisco-voip@puck.nether.net"
target=_blank>cisco-voip@puck.nether.net</A><BR><B>Subject:</B> [OSL |
CCIE_Voice] IPIPGW Sip to Sip</SPAN></P></DIV>
<DIV>
<DIV>
<P> </P>
<DIV>
<P>So, I was playing with an IPIPGW<BR><BR>CCM on one side (SIP trunk) and
CCME on the other (SIP dial-peer)... call worked, but as soon as you answered
it dropped.<BR><BR>I changed the SIP dial-peer from the IPIPGW to H.323 (no
session protocol) and RTP cuts thru fine...<BR><BR>Am I misreading something,
is SIP to SIP not supported, or is my config
retarded?<BR><BR><BR><BR>Jonathan</P></DIV></DIV></DIV></DIV></DIV></DIV>
<P> </P></DIV></DIV></DIV></DIV></DIV></BLOCKQUOTE></DIV><BR></DIV></BODY></HTML>