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<DIV><FONT face=Arial size=2>perhaps...</FONT></DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=tednugent73@gmail.com href="mailto:tednugent73@gmail.com">Ted
Nugent</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A title=lelio@uoguelph.ca
href="mailto:lelio@uoguelph.ca">Lelio Fulgenzi</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Cc:</B> <A title=greencucumber2007@yahoo.com
href="mailto:greencucumber2007@yahoo.com">Cucumber Green</A> ; <A
title=cisco-voip@puck.nether.net
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, September 20, 2007 11:44
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [cisco-voip] Antw: Analog
phones cant transfer calls on 2811acting asH323 gw to UCM 4.2 (3)</DIV>
<DIV><BR></DIV>
<DIV>Yep same here and I wondered about that also until i tried to add an ISR
FXS SCCP port to CM 4.1... its not an option like in 5.1 for example, I'm
guessing thats what they're referring to...?<BR><BR><BR> </DIV>
<DIV><SPAN class=gmail_quote>On 9/20/07, <B class=gmail_sendername>Lelio
Fulgenzi</B> <<A href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</A>>
wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<DIV bgcolor="#ffffff">
<DIV><FONT face=Arial size=2>I think the documentation is misleading...I'm
using supplementary services w/ SCCP on the VG224 with CCM 4.1(3) with no
problems.</FONT></DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV><SPAN class=e id=q_1152623e6a3bbceb_1>
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV style="BACKGROUND: #e4e4e4; FONT: 10pt arial"><B>From:</B> <A
title=tednugent73@gmail.com
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:tednugent73@gmail.com" target=_blank>Ted Nugent</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=greencucumber2007@yahoo.com
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:greencucumber2007@yahoo.com" target=_blank>Cucumber Green</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Cc:</B> <A
title=cisco-voip@puck.nether.net
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:cisco-voip@puck.nether.net"
target=_blank>cisco-voip@puck.nether.net</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, September 20, 2007
10:21 PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [cisco-voip] Antw:
Analog phones cant transfer calls on 2811acting asH323 gw to UCM 4.2
(3)</DIV>
<DIV><BR> </DIV>
<DIV>If you need features with fxs you might want to look into SCCP
for FXS controller as you would with a VG224 and h323 for and PSTN/PRI
access etc. (see the link you need CM4.2+ and a certain rev of IOS
depending on your hardware). </DIV>
<DIV> </DIV>
<DIV><A onclick="return top.js.OpenExtLink(window,event,this)"
href="http://www.cisco.com/en/US/partner/products/ps6441/products_feature_guide09186a0080483a76.html#wp1304455+"
target=_blank>http://www.cisco.com/en/US/partner/products/ps6441/products_feature_guide09186a0080483a76.html#wp1304455
</A></DIV>
<DIV> </DIV>
<DIV><A onclick="return top.js.OpenExtLink(window,event,this)"
href="http://www.cisco.com/en/US/products/ps6706/products_feature_guide09186a00806cab75.html#wp1049404"
target=_blank></A> </DIV>
<DIV> </DIV>
<DIV><BR><BR> </DIV>
<DIV><SPAN class=gmail_quote>On 9/20/07, <B
class=gmail_sendername>Cucumber Green</B> <<A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:greencucumber2007@yahoo.com"
target=_blank>greencucumber2007@yahoo.com </A>> wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<DIV>Hello!</DIV>
<DIV> </DIV>
<DIV>Yeah I have tried this without luck, I guess it only works if the
H323 gateway has a FXO</DIV>
<DIV>connection to a PBX<SPAN><BR><BR><B><I>Robert Schuknecht <<A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:rschuknecht@gmx.de"
target=_blank>rschuknecht@gmx.de</A>></I></B> wrote: </SPAN></DIV>
<BLOCKQUOTE
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid"><SPAN>Hi
Cucumber,<BR><BR>did you try to adjust the the hook-flash timer under
the voice-port?<BR>This timer has to equal on the voice-port and the
connected telephone. <BR><BR>/Robert<BR><BR></SPAN>>>>
Cucumber Green schrieb am Donnerstag, 20.
<DIV><SPAN><BR>September 2007 um 17:53 in Nachricht
3ab0adee131ef58c7afb0ea50017a410:<BR>> Analog phones can send and
receive calls from IP phones, but<BR>> when pressing the hookflash
key at the analog phone, the analog phone does <BR>> not get a dial
tone in order to transfer the call :(<BR>> <BR>> Phones are
hooked into FXS ports on a 2811 and will use G729 codec.<BR>>
<BR>> I have already checked the "Media Termination Point required"
check box in <BR>> the gateway configuration at call manager (using
unified call manager 4.2(3) <BR>> ). I thought that would do the
trick, but it didnt, any ideas?<BR>> <BR>> This is the gateway
config (IOS version 12.4(3g) spservicesK9) <BR>> <BR>> !<BR>>
version 12.4<BR>> service timestamps debug datetime msec<BR>>
service timestamps log datetime msec<BR>> service
password-encryption<BR>> !<BR>> ip subnet-zero<BR>> !<BR>>
! <BR>> ip cef<BR>> !<BR>> !<BR>> no ip domain
lookup<BR>> !<BR>> voice-card 0<BR>> no dspfarm<BR>>
!<BR>> !<BR>> !<BR>> !<BR>> interface Loopback0<BR>> ip
address X.X.X.X <A
onclick="return top.js.OpenExtLink(window,event,this)"
href="http://255.255.255.255/"
target=_blank>255.255.255.255</A><BR>> h323-gateway voip bind
srcaddr X.X.X.X<BR>> !<BR>> !<BR>> ip classless<BR>> ip
route <A onclick="return top.js.OpenExtLink(window,event,this)"
href="http://0.0.0.0/" target=_blank>0.0.0.0</A> <A
onclick="return top.js.OpenExtLink(window,event,this)"
href="http://0.0.0.0/" target=_blank>0.0.0.0</A> X.X.X.X 254<BR>>
!<BR>> !<BR>> control-plane<BR>> !<BR>> !<BR>>
!<BR>> voice-port 0/1/0 <BR>> signal loopStart<BR>> !<BR>>
voice-port 0/1/1<BR>> signal loopStart<BR>> !<BR>> dial-peer
voice 10 pots<BR>> destination-pattern 50001<BR>> port
0/1/0<BR>> !<BR>> dial-peer voice 20 pots <BR>>
destination-pattern 50002<BR>> port 0/1/1<BR>> !<BR>>
dial-peer voice 1 voip<BR>> preference 1<BR>>
destination-pattern 1...<BR>> session target ipv4:X.X.X.X<BR>>
dtmf-relay h245-alphanumeric <BR>> codec g729r8 bytes 40<BR>> ip
qos dscp cs3 media<BR>> ip qos dscp cs3 signaling<BR>> !<BR>>
dial-peer voice 4 voip<BR>> preference 2<BR>>
destination-pattern 1...<BR>> session target ipv4:X.X.X.X <BR>>
dtmf-relay h245-alphanumeric<BR>> codec g729r8 bytes 40<BR>> ip
qos dscp cs3 media<BR>> ip qos dscp cs3 signaling<BR>> !<BR>>
<BR>> <BR>> <BR>> ---------------------------------<BR>>
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