<div><STRONG>Kevin,</STRONG></div> <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><?xml:namespace prefix = o ns = "urn:schemas-microsoft-com:office:office" /><o:p> </o:p></SPAN></div> <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"> <o:p></o:p></SPAN></div> <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Welcome back and thanks for your reply, <o:p></o:p></SPAN></div> <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></div> <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">1-I want to ask you about the G711alaw I think it is need 66 kbps but as you know that the Voice channel in E1 is 64 Kbps only so is this difference makes any problem? If I change the G711alaw to another Codec like G729 is this change solves the problem?<o:p></o:p></SPAN></div> <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></div> <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">2-I have Firewall and put
the CCM at server zone and the IP Phone and the VGW in other zone.<o:p></o:p></SPAN></div> <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Is there is any problem in this design? Is it preferred to put the CCM and the VGW <SPAN style="mso-spacerun: yes"> </SPAN>and the IP Phone at the same Subnet (zone)?<o:p></o:p></SPAN></div> <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p></o:p></SPAN> </div> <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p>Thanks</o:p></SPAN></div> <div><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p>Waleed </o:p></SPAN></div> <div><BR><B><I>Kevin Thorngren <kthorngr@cisco.com></I></B> wrote:</div> <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">Sorry for the delay, I was busy the past couple of days and did not <BR>have much time for email. You should reply to all that way someone <BR>else on the email list might respond and answer your remaining
<BR>questions.<BR><BR>G711alaw is fine to use with E1. The serial interface config is <BR>specific to your environment so I wouldn't be able to tell you if <BR>something is needed or not. If it is working I would leave it as is. <BR>I don't believe there is anything there that would cause one way audio <BR>issues.<BR><BR>Kevin<BR>On Mar 10, 2006, at 3:23 PM, new look wrote:<BR><BR>> Dear Kevin,<BR>> I'm still waiting your reply Thanks.<BR>> <BR>> Kevin,<BR>> Thanks a lot for your fast response,<BR>> First, the GW and the IP Phone are in the same subnet.<BR>> Second, what about the serial interface config. And the codec type?<BR>> Thanks <BR>> <BR>><BR>> Kevin Thorngren <KTHORNGR@CISCO.COM>wrote:<BR>>> The first place to look is to make sure that you don't have any IP<BR>>> connectivity issues during the one way audio. Although your diagram<BR>>> shows a pretty simple network but the phones may be on a
different<BR>>> subnet from the GW. You can hit the "i" or "?" button twice quickly<BR>>> while on the call. This will show you the call stats along the RxCnt.<BR>>> If the RxCnt is increasing then the RTP stream is getting to the phone<BR>>> but there is apparently blank audio if you are not hearing anything.<BR>>> If the counter is not increasing then the problem is some sort of IP<BR>>> connectivity issue between the GW and the Phone.<BR>>><BR>>> You can use the "show voice dsp" command to see if there are any DSPs<BR>>> that are having transmit or receive issues. Just look in the<BR>>> TX/RX-PAK-CNT columns.<BR>>><BR>>> It doesn't look like dial peers 2,3,4 or 102 are needed. There are no<BR>>> voice ports assigned to 2,3 or 4 and no session target assigned to <BR>>> 102.<BR>>><BR>>> Kevin<BR>>> On Mar 8, 2006, at 6:37 PM, new look wrote:<BR>>><BR>>> > Dear
All,<BR>>> > I have a performance problem in my IP Phone system with the E1 calls<BR>>> > (Some times I can't hear the person I called him), my system already<BR>>> > configured before. (My network Diagram is attached)<BR>>> > So first can any one help me to know if the existing configuration <BR>>> has<BR>>> > any wrong or unneeded commands? (The configuration of the VGW is<BR>>> > attached)<BR>>> > Note: My IP Phone DN#'s are 11XX<BR>>> > <BR>>> ><BR>>> > 1-I want to know if the interface Serial0/0/0:15 configurations have<BR>>> > any wrong commands or unneeded commands?<BR>>> > 2-I want to know if I need dial-peers 2, 3 and 4, or not?<BR>>> > 3-I want to know if I need dial-peers 102, why?<BR>>> > 4-Is the Codec g711ulaw is right codec for E1 calls or I need to use<BR>>> > another Codec? which one?<BR>>> > 5-is any
one has the URL for Cisco example for this solution ?<BR>>> ><BR>>> > Yahoo! Mail<BR>>> > Bring photos to life! New PhotoMail makes sharing a breeze.isdn<BR>>> > switch-type primary-net5<BR>>> > !<BR>>> > voice-card 0<BR>>> > no dspfarm<BR>>> > !<BR>>> > voice translation-rule 1<BR>>> > rule 1 /^84/ /11/<BR>>> > !<BR>>> > voice translation-profile incom<BR>>> > translate called 1<BR>>> > !<BR>>> > controller E1 0/0/0<BR>>> > framing NO-CRC4<BR>>> > pri-group timeslots 1-31<BR>>> > !<BR>>> > translation-rule 1<BR>>> > !<BR>>> > !<BR>>> > interface Serial0/0/0:15<BR>>> > no ip address<BR>>> > isdn switch-type primary-net5<BR>>> > isdn overlap-receiving<BR>>> > isdn not-end-to-end 64<BR>>> > isdn incoming-voice voice<BR>>> > isdn
guard-timer 20000 on-expiry accept<BR>>> > isdn map address * plan unknown type unknown<BR>>> > isdn map address .* plan unknown type unknown<BR>>> > isdn map address transparent<BR>>> > isdn T309-enable<BR>>> > isdn T306 400000<BR>>> > isdn sending-complete<BR>>> > isdn incoming alerting add-PI<BR>>> > no isdn gtd<BR>>> > no cdp enable<BR>>> > !<BR>>> > voice-port 0/0/0:15<BR>>> > !<BR>>> > dial-peer voice 1 pots<BR>>> > translation-profile incoming incom<BR>>> > destination-pattern .T<BR>>> > direct-inward-dial<BR>>> > port 0/0/0:15<BR>>> > !<BR>>> > dial-peer voice 2 pots<BR>>> > preference 1<BR>>> > destination-pattern .T<BR>>> > !<BR>>> > dial-peer voice 3 pots<BR>>> > preference 2<BR>>> > destination-pattern .T<BR>>> > !<BR>>> > dial-peer
voice 4 pots<BR>>> > preference 3<BR>>> > destination-pattern .T<BR>>> > !<BR>>> > dial-peer voice 100 voip<BR>>> > destination-pattern 1...<BR>>> > session target ipv4:172.16.80.24<BR>>> > dtmf-relay h245-alphanumeric<BR>>> > codec g711alaw<BR>>> > !<BR>>> > dial-peer voice 102 voip<BR>>> > dtmf-relay h245-alphanumeric<BR>>> > codec g711ulaw Yahoo! Mail<BR>> Use Photomail to share photos without annoying attachments.</BLOCKQUOTE><BR><p>
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