Re: [OSL | CCIE_Voice] Trancoding

2012-09-19 Thread William Affeldt
It looks like you are missing the toll fraud prevention. Try the commands 
below.   


Voice service VoIP 
Allow connections h323 to sip 
Allow connections sip to h322 
Allow connections sip to sip
Allow connections h323 to h323

Sent from my iPhone

On Sep 19, 2012, at 5:42 AM, Chrysostomos Christofi ch.christ...@logicom.net 
wrote:

 Hi Justin
 
 I have removed the class from the inbound dial peers and the results are the 
 same
 The transcoding does not get  involved
 
 Any ideas?
 
 -Original Message-
 From: Justin McIntyre [mailto:justin.mcint...@blackbox.com] 
 Sent: Τετάρτη, 19 Σεπτεμβρίου 2012 3:25 μμ
 To: Chrysostomos Christofi
 Cc: ccie_voice@onlinestudylist.com
 Subject: Re: Trancoding
 
 Try taking the Voice Class codec out of the inbound dial-peer that you have 
 here.  You should see then then that your transcoder will start to be 
 invoked.  I believe what is happening to you is that you are sending a 
 request over to use 729, and then the internal call to CUE is requesting 
 711u.  The transcoder will only be invoked if the inbound dial peer does not 
 support the list of codecs.  In your case, the inbound dial-peer does support 
 both codecs (voice-class codec 1), so the transcoder is not being invoked.  
 Simply remove the command voice-class codec1 from your dial-peer listed below 
 and see what happens.
 
 Thanks,
 
 Justin McIntyre
 
 On Sep 19, 2012, at 8:19 AM, Chrysostomos Christofi 
 ch.christ...@logicom.net wrote:
 
 Hi
 
 I have
 
 !
 voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g711alaw
 codec preference 3 g729r8
 !
 dial-peer voice 100 voip
 preference 10
 destination-pattern 1...
 session target ipv4:CUCM IP
 incoming called-number .
 voice-class codec 1
 dtmf-relay h245-alphanumeric
 no vad
 
 !
 
 But the transcoding doesn't being involved
 
 Regards
 
 
 
 -Original Message-
 From: ccie_voice-boun...@onlinestudylist.com 
 [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Justin 
 McIntyre
 Sent: Τετάρτη, 19 Σεπτεμβρίου 2012 2:44 μμ
 To: ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Trancoding
 
 Do you have a voice-class codec applied to your in bound dial-peer?   This 
 will keep the transcoder from being invoked.
 
 Thanks,
 
 Justin McIntyre
 
 On Sep 19, 2012, at 6:45 AM, ccie_voice-requ...@onlinestudylist.com 
 ccie_voice-requ...@onlinestudylist.com wrote:
 
 Send CCIE_Voice mailing list submissions to
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 When replying, please edit your Subject line so it is more specific 
 than Re: Contents of CCIE_Voice digest...
 
 
 Today's Topics:
 
 1. Re: Trancoding (Bill Lake)
 2. Re: Trancoding (Chrysostomos Christofi)
 
 
 -
 -
 
 Message: 1
 Date: Wed, 19 Sep 2012 05:28:30 -0500
 From: Bill Lake whl...@gmail.com
 To: Chrysostomos Christofi ch.christ...@logicom.net
 Cc: Online Study \(ccie_voice@onlinestudylist.com\)
  ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Trancoding
 Message-ID:
 
 CADpb93M_qAh6Kg2840VRctLMxyWa9drqG8BHk2iOkgUfm4=1...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-7
 
 Can you give us more information, you say VM does not work but does 
 it work for local callers?  does it work for PSTN callers?  Or do you 
 not have PSTN callers?
 
 Can you dial the VM directly?
 
 What are the results of show sccp conn when you do this?
 
 We just don't have any information but a partial config and it 
 doesn't work.  Not enough to go on.
 
 
 
 Now this is just a wild guess but I seem to remember someone else 
 having an issue like this and they resolved it by using the loopback 
 and not a sub interface.  Of course I could be completely wrong as 
 this is just something that is in my foggy memory but don't think it could 
 hurt to try.
 
 
 
 
 On another note, why do you have
 
 
 interface GigabitEthernet0/0.100
 
 encapsulation dot1Q 100
 
 ip address 10.4.2.1 255.255.255.0
 
 h323-gateway voip interface
 
 h323-gateway voip h323-id  cme
 h323-gateway voip bind srcaddr 10.4.2.1
 
 you appear to be setting up for a Gatekeeper but do not have the rest 
 of the commands.  If this is just an H323 gateway you do not need 
 h323-gateway voip interface or h323-gateway voip h323-id cme
 
 
 On Wed, Sep 19, 2012 at 2:45 AM, Chrysostomos Christofi  
 ch.christ...@logicom.net wrote:
 
 Hi
 
 ** **
 
 Any update?
 
 ** **
 
 Regards
 
 ** **
 
 ** **
 
 *From:* Chrysostomos Christofi
 *Sent:* ?, 18 ??? 2012 1:23 ??
 *To:* 'ccie_voice-boun...@onlinestudylist.com'
 *Subject:* Trancoding
 
 ** **
 
 Hi to all
 
 ** **
 
 I need your input pls 

[OSL | CCIE_Voice] CME to CCM Presence

2012-09-14 Thread William Affeldt
Hey Experts,
 
I am having an issue with CME to CM presence. The CME can see the CM presence 
but the CM can't see the CME presence. It never even send the message to CM. 
The debug presence and important config is below. Any help would be greatly 
appreciated.
 
Sep 15 05:27:43.141: //PRESENCE:[0]:/presence_line_update: SCCP, no update
Sep 15 05:27:43.145: //PRESENCE:[0]:/presence_sip_line_update: SIP nothing to 
update
 
  
!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
  bind control source-interface Loopback0
  bind media source-interface Loopback0
!
!
!
presence
presence call-list
server 10.10.210.10
watcher all
allow subscribe
!
gateway
timer receive-rtp 1200
!
sip-ua
mwi-server ipv4:10.10.115.2 expires 3600 port 5060 transport udp unsolicited
presence enable
!
!
telephony-service
sdspfarm units 2
sdspfarm transcode sessions 4
sdspfarm tag 1 sitectrans
no auto-reg-ephone
authentication credential admin cisco
em logout 0:0 0:0 0:0
max-ephones 10
max-dn 20
ip source-address 10.10.110.3 port 2000
system message Your current options
url services http://10.10.115.2/voiceview/common/login.do
url authentication http://10.10.110.3/CCMCIP/authenticate.asp
time-format 24
date-format yy-mm-dd
voicemail 4600
max-conferences 8 gain -6
web admin system name admin password cisco
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp 7960 Sep 15 2012 04:14:41
!
!
ephone-dn  1  octo-line
number 4001 no-reg primary
description +442077964001
name SiteCPhone1
allow watch
call-forward busy 4600
call-forward noan 4600 timeout 10
!
!
ephone-dn  2  octo-line
number 4002
description +442077964002
name SiteCPhone2
allow watch
call-forward busy 4600
call-forward noan 4600 timeout 10
!
!
ephone-dn  5  octo-line
number *4002
call-forward all 4600
!
!
ephone  1
mac-address 0021.A086.8796
username sitecphn1
type 7962
button  1:1
pin 12345
!
!
!
ephone  2
mac-address E8BA.70FA.A942
username tjefferson
presence call-list
blf-speed-dial 1 2002 label 2002
type 7975
button  1:2 6m5
pin 12345
!
!
 
 
 
 
 
 
 
 



From: Bill Lake whl...@gmail.com
To: John John john_ccie2...@yahoo.com 
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com 
Sent: Friday, September 14, 2012 3:11 PM
Subject: Re: [OSL | CCIE_Voice] sip trunk


Sure you can, you know that any call incoming from Site A must have ANI of 
332211 and any call incoming from Site B must have 33 so you just need to 
enforce that and you could do that with a incoming voice translation rule.  

So no matter what the customer sends you can change that to what you want it to 
be
 
We do this all the time in the work books

voice translation-rule 1 
rule 1 /^3...$/ /408387\0/ 

this expands 3001 to 4083873001 

but you could also do

rule 1 /^3...$/ /4001/  

This would take any 3XXX number and change it to 4001, why you ask, just think 
you have 800 people calling out from your company call center and this prevents 
800 different numbers from going out. Everyone you call gets 4001.  not 
3001-3799 or something like that.

You get the idea?




On Fri, Sep 14, 2012 at 4:35 PM, John John john_ccie2...@yahoo.com wrote:

Dear Pavan,
 
Thanks for your feedback,But my question is even if i implement cor still I 
can't control that because I can't control what customer A and B and in that 
case if customer A sent me the call with ANI for cutomer B then customer B 
dial-peer will be match and the call will not be blocked.
 
but during my google search I found some greating feature which is incoming 
uri based on the IP so anybody working in that feature 

  
From: Pavan pav.c...@gmail.com
To: John John john_ccie2...@yahoo.com 
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com 
Sent: Friday, September 14, 2012 6:43 AM
Subject: Re: [OSL | CCIE_Voice] sip trunk


I think cor can offer a good solution for this. Assign a separate cor group to 
each dialpeer say cust1 and cust2 in both inbound and outbound directions

-Pavan

On Sep 13, 2012, at 17:45, John John john_ccie2...@yahoo.com wrote:


Dear All,

 We have two sipt trunk for 2 comapany:-

company A - DID range 332211XX
Company B - DID range 33XX

and each company has own PBX,Company A  has AVAYA and company B has cisco 
Call manager.

and they have sip trunk to my gateway where is the E1 is connected.

in my gateway there is two dial peer :

dial-peer voice 1 voip
 description ## Customer-A ##
 answer-address 332211..
 destination-pattern 332211..
 modem passthrough nse codec g711ulaw
 session protocol sipv2
 session target ipv4:192.168.10.10
 
dial-peer voice 2 voip
 description ## Customer-B ##
 answer-address 33..
 destination-pattern 33..
 modem passthrough nse codec g711ulaw
 session protocol sipv2
 session target ipv4:192.168.20.10
 

So in my case if customer A send me the ANI as 3322 which is for 

Re: [OSL | CCIE_Voice] sccp issue with the router

2012-09-10 Thread William Affeldt
5 or greater should be fine. 

Sent from my iPhone

On Sep 10, 2012, at 2:09 PM, otunola Akerele otunola.aker...@gmail.com wrote:

 hi guys, am using the proctor lab and the hq-rtr ios is Version 12.4(15)T8 
 and not giving me version 7 while trying to register with  the cucm. will  
 the sccp register??
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Re: [OSL | CCIE_Voice] Is it possible to prevent users from dialing VM Pilot

2012-09-04 Thread William Affeldt
Check the hunt pilot. What partition is it in? 

Sent from my iPhone

On Sep 4, 2012, at 8:03 AM, Pavan K pav.c...@gmail.com wrote:

 UCM 8.6.2 / Unity Connection 8.6.2
 
 Is there a way to prevent users from dialing the Voicemail pilot directly ?
 
 
 I tried the following but it doesn't work
 (1) Place the Unity Route Pattern in a Partition (lets say Voicemail)
 (2) Configure the CSS on the Voicemail Pilot to include the Voicemail 
 Partition
 (3) Device/Line CSS on the Phone does not have the Voicemail pilot in it
 
 It seems to always use the Line/Device CSS to call into the Voicemail
 System when
 the Voicemail button is pressed on the phone (instead of using the CSS
 configured in Voicemail Pilot / Profile assigned to the device).
 
 
 Forwarded calls get routed to Voicemail as the CF CSS is set to use
 With Configured CSS and the CSS for Forwarding has the Voicemail CSS
 
 -- 
 - Pavan
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Re: [OSL | CCIE_Voice] dead air when calling CUE vmail gui doesn't work

2012-07-05 Thread William Affeldt
I noticed you were missing the static route pointing to the service module. 
Also, make sure you have a default gateway on the service module pointing to 
the IP unnumbered interface. Finally if the ip unnumbered interface is a loop 
back make sure it has the ip ospf network point to point command on it. I think 
that is the command you can question mark it if it is not. 




Sent from my iPhone

On Jul 4, 2012, at 8:21 PM, Krishna vinayak_...@yahoo.com wrote:

 hi folks,
 
 I couldn't able to understand why the CUE giving me the dead air though after 
 the configuration is absolutely correct with the right codecs. when i pressed 
 the vmail button on the phone, it connects to the vmail number but i cannot 
 hear anything. 
 
 And, also i couldn't access web gui for the cue even after providing all the 
 right info such as ip http server, ip http path, ip http auth local.. the web 
 browser sits there forever with no output...
 
 does anyone experienced the same problem as i am??? your advice on this 
 matter is much appreciated.
 
 
 thank you
 krishna.
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Re: [OSL | CCIE_Voice] Proctor Labs issue?

2012-06-08 Thread William Affeldt
I had the same issue yesterday. It only happened on the CCM though not CME. 

Sent from my iPhone

On Jun 8, 2012, at 10:51 AM, Dan Quinlan (daquinla) daqui...@cisco.com 
wrote:

 All,
 
 The past two times I've used ProctorLabs for rack time, I've seen my local 
 (local to me) IP Phones deregister / reregister every few minutes and they 
 usually don't come all the way back (DN's missing, etc).  I haven't changed 
 my hardware VPN connectivity at all. Has anyone else seen issues lately?
 
 Tia 
 
 DQ
 d...@cisco.com
 
 Sent from my iPhone
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[OSL | CCIE_Voice] Proctorlabs issues?

2012-05-29 Thread William Affeldt
Anyone else having trouble connecting to proctorlabs.com?___
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Re: [OSL | CCIE_Voice] Issue with proctorlabs site

2012-05-22 Thread William Affeldt
The current version of Java is 7.4. I am going to guess that you haven't 
updated in a while. 

Sent from my iPhone

On May 22, 2012, at 5:10 AM, Anthony Sequeira anthony.seque...@stormwind.com 
wrote:

 Just as an FYI and I hope this can help - I am at IE version 9.0.8112.16421 
 and Java 1.6 with no issues. The OS is Windows 7 Home Premium 64-bit
  
 From: ccie_voice-boun...@onlinestudylist.com 
 [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of William Affeldt
 Sent: Monday, May 21, 2012 9:12 PM
 To: ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] Issue with proctorlabs site
  
 Hey Gents,
  
 I updated my IE and java today and now the proctorlabs site won't load the 
 login prompt or show the images of the different lab options. Also, an update 
 on my MAC did the same thing. Has anyone fixed this yet or ran into the same 
 problem? Any help would be greatly appreciated. 
  
  
___
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Re: [OSL | CCIE_Voice] Issue with proctorlabs site

2012-05-22 Thread William Affeldt

LOL, I feel stupid. It was the flash player. I am good now. 
 


 From: Anthony Sequeira anthony.seque...@stormwind.com
To: 'William Affeldt' william.affe...@yahoo.com; 
'ccie_voice@onlinestudylist.com' ccie_voice@onlinestudylist.com 
Sent: Tuesday, May 22, 2012 5:10 AM
Subject: RE: [OSL | CCIE_Voice] Issue with proctorlabs site
  

Just as an FYI and I hope this can help - I am at IE version 9.0.8112.16421 and 
Java 1.6 with no issues. The OS is Windows 7 Home Premium 64-bit
 
From:ccie_voice-boun...@onlinestudylist.com 
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of William Affeldt
Sent: Monday, May 21, 2012 9:12 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Issue with proctorlabs site
 
Hey Gents,
 
I updated my IE and java today and now the proctorlabs site won't load the 
login prompt or show the images of the different lab options. Also, an update 
on my MAC did the same thing. Has anyone fixed this yet or ran into the same 
problem? Any help would be greatly appreciated. ___
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[OSL | CCIE_Voice] Issue with proctorlabs site

2012-05-21 Thread William Affeldt
Hey Gents,

I updated my IE and java today and now the proctorlabs site won't load the 
login prompt or show the images of the different lab options. Also, an update 
on my MAC did the same thing. Has anyone fixed this yet or ran into the same 
problem? Any help would be greatly appreciated. ___
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Re: [OSL | CCIE_Voice] isdn switch type

2012-04-10 Thread William Affeldt
Euro



Sent from my iPhone

On Apr 10, 2012, at 5:29 AM, Ken Wyan kew...@gmail.com wrote:

 I want to configure a 2811 Router as a MGCP Gateway using CUCM GUI.
  
 I can't find isdn switch-type primary-net5  under CUCM GUI mgcp endpoint 
 configuration page.
  
 which one is equivalent to primary-net5 available in IOS ?
  
 Thanks
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Re: [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration

2012-02-04 Thread William Affeldt
I am currently having the same problem. I have been troubleshooting for a hour 
now. It forwards to the correct VM box and you can even play the message back 
to your self after you record it. It just never makes it to the mailbox. 




From: Edgar Feliz ejzi...@gmail.com
To: ccie_voice@onlinestudylist.com 
Sent: Saturday, February 4, 2012 10:15 AM
Subject: [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration


I am currently working on new lab 5 and I have my CUCM-CUC integration working, 
for voicemail left by SA  SB to SB phone and SB  SA Phone I get MWI 
both directions. But for PSTN or SC While I can leave a VM MWI does not work 
and the VM does not show up when I check the inbox for either SA or SB phones 
for VM left from PSTN or SC. I have looked at the SIP trunk setting and do not 
see anything there CSS/PTs all look correct any Ideas? 

Thanks

E
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Re: [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration

2012-02-04 Thread William Affeldt
You are having one way audio issues then. 

Sent from my iPhone

On Feb 4, 2012, at 11:27 AM, Edgar Feliz ejzi...@gmail.com wrote:

 I don't hear the message when I press # for more options from SC or PSTN 
 nothing is happening but I am getting the options from SA/SB
 
 E 
 
 On Sat, Feb 4, 2012 at 1:44 PM, William Affeldt william.affe...@yahoo.com 
 wrote:
 I am currently having the same problem. I have been troubleshooting for a 
 hour now. It forwards to the correct VM box and you can even play the message 
 back to your self after you record it. It just never makes it to the mailbox. 
 
 From: Edgar Feliz ejzi...@gmail.com
 To: ccie_voice@onlinestudylist.com 
 Sent: Saturday, February 4, 2012 10:15 AM
 Subject: [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration
 
 I am currently working on new lab 5 and I have my CUCM-CUC integration 
 working, for voicemail left by SA  SB to SB phone and SB  SA Phone I get 
 MWI both directions. But for PSTN or SC While I can leave a VM MWI does not 
 work and the VM does not show up when I check the inbox for either SA or SB 
 phones for VM left from PSTN or SC. I have looked at the SIP trunk setting 
 and do not see anything there CSS/PTs all look correct any Ideas?
 
 Thanks
 
 E
 
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Re: [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration

2012-02-04 Thread William Affeldt
So I am good then? 

Sent from my iPhone

On Feb 4, 2012, at 11:59 AM, Vik Malhi vma...@ipexpert.com wrote:

 Edgar- make sure that you do not have one way cRTP. Or there is any MTP being 
 used no sccp/sccp.
 
 Bill-  there is a CUC bug when you leave a VM  and can press # and hear your 
 message, but this message never gets sent to the mail box (from specific ip 
 addresses). In this case you have to just rely on VM/MWI from an extension 
 /gateway that does work.
 
 
 
 -- 
 Vik Malhi – CCIE #13890
 Managing Partner / Instructor - IPexpert, Inc.
 Mailto: vma...@ipexpert.com
 Telephone: +1.810.326.1444
 Fax: +1.810.454.0130
 Live Assistance, Please visit: www.ipexpert.com/chat
 http://www.ipexpert.com/chat
 
 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, 
 Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE 
 (RS, Voice, Wireless, Security  Service Provider) certification(s) with 
 training locations throughout the United States, Europe, South Asia and 
 Australia. Be sure to visit our online communities at 
 www.ipexpert.com/communities http://www.ipexpert.com/communities  and our 
 public website at www.ipexpert.com http://www.ipexpert.com/
 
 
 On Feb 4, 2012, at 11:42 AM, William Affeldt william.affe...@yahoo.com 
 wrote:
 
 You are having one way audio issues then. 
 
 Sent from my iPhone
 
 On Feb 4, 2012, at 11:27 AM, Edgar Feliz ejzi...@gmail.com wrote:
 
 I don't hear the message when I press # for more options from SC or PSTN 
 nothing is happening but I am getting the options from SA/SB
 
 E 
 
 On Sat, Feb 4, 2012 at 1:44 PM, William Affeldt william.affe...@yahoo.com 
 wrote:
 I am currently having the same problem. I have been troubleshooting for a 
 hour now. It forwards to the correct VM box and you can even play the 
 message back to your self after you record it. It just never makes it to 
 the mailbox. 
 
 From: Edgar Feliz ejzi...@gmail.com
 To: ccie_voice@onlinestudylist.com 
 Sent: Saturday, February 4, 2012 10:15 AM
 Subject: [OSL | CCIE_Voice] New lab 5 CUCM/CUC SIP Integration
 
 I am currently working on new lab 5 and I have my CUCM-CUC integration 
 working, for voicemail left by SA  SB to SB phone and SB  SA Phone I get 
 MWI both directions. But for PSTN or SC While I can leave a VM MWI does not 
 work and the VM does not show up when I check the inbox for either SA or SB 
 phones for VM left from PSTN or SC. I have looked at the SIP trunk setting 
 and do not see anything there CSS/PTs all look correct any Ideas?
 
 Thanks
 
 E
 
 ___
 For more information regarding industry leading CCIE Lab training, please 
 visit www.ipexpert.com
 
 Are you a CCNP or CCIE and looking for a job? Check out 
 www.PlatinumPlacement.com
 
 
 ___
 For more information regarding industry leading CCIE Lab training, please 
 visit www.ipexpert.com
 
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 www.PlatinumPlacement.com
___
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[OSL | CCIE_Voice] BACD Not Invoking service

2012-02-04 Thread William Affeldt
I configured BACD today and could not get it working. The dial-peer never 
actually invokes the application. It does hit the correct dial-peer in and out 
though. Below is my exact configuration. Any help would be greatly appreciated. 
 
 
 
application
service app-b-acd-aa
param voice-mail 5003
paramspace english index 1
param max-time-call-retry 700
param service-name app-b-acd
param number-of-hunt-grps 1
paramspace english language en
param handoff-string app-b-acd-aa
param dial-by-extension-option 1
param max-time-vm-retry 2
paramspace english location flash:bacdprompts/
param aa-pilot 4110
param second-greeting-time 60
param welcome-prompt _bacd_welcome.au
param queue-manager-debugs 1000
param call-retry-timer 15
 !
service app-b-acd
param queue-len 15
param number-of-hunt-grps 1
param aa-hunt2 
param queue-manager-debugs 1
 !
!
!
dial-peer voice 222 voip
 service app-b-acd-aa
 destination-pattern 4110
 session target ipv4:10.10.110.3
 incoming called-number 4110
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!




From: Vik Malhi vma...@ipexpert.com
To: Ashwani ash_r...@hotmail.com 
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com 
Sent: Saturday, February 4, 2012 12:45 PM
Subject: Re: [OSL | CCIE_Voice] Reorder tone while + dialing from 
directories(missed, received)

From SIP phones calls from the directory are sent digit by digit. This is in 
contrast to sccp phones which send digits en bloc (as opposed to digit by 
digit).

A route pattern such as : \+! marked as urgent priority would cause a call from 
the directory from a sip phone to fail. Since the plus would match and since 
the urgent priority has been selected the call would get sent to the gateway 
with just a + (which would be stripped in the case of an h323 gateway).

Remove and plus route patterns marked as urgent priority- if you want to avoid 
inter digit timeout create route patterns without the ! and define the exact 
number of digits (x's).



-- 
Vik Malhi – CCIE #13890
Managing Partner / Instructor - IPexpert, Inc.
Mailto: vma...@ipexpert.com
Telephone: +1.810.326.1444
Fax: +1.810.454.0130
Live Assistance, Please visit: www.ipexpert.com/chat
http://www.ipexpert.com/chat

IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio 
Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, 
Voice, Wireless, Security  Service Provider) certification(s) with training 
locations throughout the United States, Europe, South Asia and Australia. Be 
sure to visit our online communities at www.ipexpert.com/communities 
http://www.ipexpert.com/communities  and our public website at 
www.ipexpert.com http://www.ipexpert.com/


On Feb 4, 2012, at 12:27 PM, Ashwani ash_r...@hotmail.com wrote:

 Hello Everyone,
 
 I am having issue + dialing (missed, received) from HQ, BR1 ( 7961 SIP ) 
 phones.  I can dial from 7961 SCCP phones , other 7965 and 7962 phones.  Can 
 someone please explain me why I am not able to dial from SIP phones?
 
 Here are the testing I have done so far..
 
 1) Dialing from PSTN phone to 7961 SIP phones  Call works.
 2) Dialing from 7961 SIP phones to PSTN numbers  Call works
 3) Dialing from missed/received of 7961 SCCP and other model phones  Call 
 works
 4) Dialing from missed/received of 7961 SIP  Doesnt work
 
 Any help will be appreciated.
 
 Thanks,
 Ashwani
 ___
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 visit www.ipexpert.com
 
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 www.PlatinumPlacement.com
___
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www.ipexpert.com

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Re: [OSL | CCIE_Voice] BACD Not Invoking service

2012-02-04 Thread William Affeldt
I don't see it. I have what you said. 

Sent from my iPhone

On Feb 4, 2012, at 5:07 PM, Jason Langenfeld jlangenf...@prosysis.com wrote:

 Wrong service name…
 
 
 application
 service app-b-acd-aa
 
 
 dial-peer voice 222 voip
 service app-b-acd-aa
 
 
 service app-b-acd
 
 Thanks,
 Jason Langenfeld  |  Systems Engineer
 Prosys
 6550 Carothers Parkway, Suite 420
 Franklin, TN 37067
 O 615.435.1540
 Email jlangenf...@prosysis.com
 
 
 From: William Affeldt 
 william.affe...@yahoo.commailto:william.affe...@yahoo.com
 Reply-To: William Affeldt 
 william.affe...@yahoo.commailto:william.affe...@yahoo.com
 Date: Sat, 4 Feb 2012 16:43:36 -0800
 To: Vik Malhi vma...@ipexpert.commailto:vma...@ipexpert.com, Ashwani 
 ash_r...@hotmail.commailto:ash_r...@hotmail.com
 Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com 
 ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice] BACD Not Invoking service
 
 I configured BACD today and could not get it working. The dial-peer never 
 actually invokes the application. It does hit the correct dial-peer in and 
 out though. Below is my exact configuration. Any help would be greatly 
 appreciated.
 
 
 
 application
 service app-b-acd-aa
 param voice-mail 5003
 paramspace english index 1
 param max-time-call-retry 700
 param service-name app-b-acd
 param number-of-hunt-grps 1
 paramspace english language en
 param handoff-string app-b-acd-aa
 param dial-by-extension-option 1
 param max-time-vm-retry 2
 paramspace english location flash:bacdprompts/
 param aa-pilot 4110
 param second-greeting-time 60
 param welcome-prompt _bacd_welcome.au
 param queue-manager-debugs 1000
 param call-retry-timer 15
 !
 service app-b-acd
 param queue-len 15
 param number-of-hunt-grps 1
 param aa-hunt2 
 param queue-manager-debugs 1
 !
 !
 !
 dial-peer voice 222 voip
 service app-b-acd-aa
 destination-pattern 4110
 session target ipv4:10.10.110.3
 incoming called-number 4110
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
 !
 
 From: Vik Malhi vma...@ipexpert.commailto:vma...@ipexpert.com
 To: Ashwani ash_r...@hotmail.commailto:ash_r...@hotmail.com
 Cc: ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com 
 ccie_voice@onlinestudylist.commailto:ccie_voice@onlinestudylist.com
 Sent: Saturday, February 4, 2012 12:45 PM
 Subject: Re: [OSL | CCIE_Voice] Reorder tone while + dialing from 
 directories(missed, received)
 
 From SIP phones calls from the directory are sent digit by digit. This is in 
 contrast to sccp phones which send digits en bloc (as opposed to digit by 
 digit).
 
 A route pattern such as : \+! marked as urgent priority would cause a call 
 from the directory from a sip phone to fail. Since the plus would match and 
 since the urgent priority has been selected the call would get sent to the 
 gateway with just a + (which would be stripped in the case of an h323 
 gateway).
 
 Remove and plus route patterns marked as urgent priority- if you want to 
 avoid inter digit timeout create route patterns without the ! and define 
 the exact number of digits (x's).
 
 
 
 --
 Vik Malhi – CCIE #13890
 Managing Partner / Instructor - IPexpert, Inc.
 Mailto: vma...@ipexpert.commailto:vma...@ipexpert.com
 Telephone: +1.810.326.1444
 Fax: +1.810.454.0130
 Live Assistance, Please visit: www.ipexpert.com/chat
 http://www.ipexpert.com/chat
 
 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, 
 Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE 
 (RS, Voice, Wireless, Security  Service Provider) certification(s) with 
 training locations throughout the United States, Europe, South Asia and 
 Australia. Be sure to visit our online communities at 
 www.ipexpert.com/communitieshttp://www.ipexpert.com/communities 
 http://www.ipexpert.com/communities  and our public website at 
 www.ipexpert.comhttp://www.ipexpert.com/ http://www.ipexpert.com/
 
 
 On Feb 4, 2012, at 12:27 PM, Ashwani 
 ash_r...@hotmail.commailto:ash_r...@hotmail.com wrote:
 
 Hello Everyone,
 
 I am having issue + dialing (missed, received) from HQ, BR1 ( 7961 SIP ) 
 phones.  I can dial from 7961 SCCP phones , other 7965 and 7962 phones.  Can 
 someone please explain me why I am not able to dial from SIP phones?
 
 Here are the testing I have done so far..
 
 1) Dialing from PSTN phone to 7961 SIP phones  Call works.
 2) Dialing from 7961 SIP phones to PSTN numbers  Call works
 3) Dialing from missed/received of 7961 SCCP and other model phones  Call 
 works
 4) Dialing from missed/received of 7961 SIP  Doesnt work
 
 Any help will be appreciated.
 
 Thanks,
 Ashwani
 ___
 For more information regarding industry leading CCIE Lab training, please 
 visit www.ipexpert.com
 
 Are you a CCNP or CCIE and looking for a job? Check out 
 www.PlatinumPlacement.comhttp://www.platinumplacement.com/
 ___
 For more information

Re: [OSL | CCIE_Voice] BACD Not Invoking service

2012-02-04 Thread William Affeldt
No, we removed that portion to make sure it wasn't causing issues. My config 
worked fine on another device. 

Sent from my iPhone

On Feb 4, 2012, at 8:26 PM, Baktha Muralidharan muralic...@gmail.com wrote:

 Hi William
 
 - did you mean to configure a drop-through B-ACD? then you need 
param drop-through-option 1
 - You probably want param aa-hunt1  instead of param aa-hunt2 
 - make sure moh is enabled and working
 - if it still doesn't work, please turn ON debug voice application vxml and 
 post the debug capture.
 
 thanks,
 /Baktha
 
 
 From: William Affeldt william.affe...@yahoo.com
 To: Vik Malhi vma...@ipexpert.com, Ashwani ash_r...@hotmail.com
 Cc: ccie_voice@onlinestudylist.com  ccie_voice@onlinestudylist.com
 Subject: [OSL | CCIE_Voice]   BACD Not Invoking service
 Message-ID:
1328402616.84156.yahoomail...@web114616.mail.gq1.yahoo.com
 Content-Type: text/plain; charset=utf-8
 
 I configured BACD today and could not get it working. The dial-peer never 
 actually invokes the application. It does hit the correct dial-peer in and 
 out though. Below is my exact configuration. Any help would be greatly 
 appreciated.
 ?
 ?
 ?
 application
 service app-b-acd-aa
 param voice-mail 5003
 paramspace english index 1
 param max-time-call-retry 700
 param service-name app-b-acd
 param number-of-hunt-grps 1
 paramspace english language en
 param handoff-string app-b-acd-aa
 param dial-by-extension-option 1
 param max-time-vm-retry 2
 paramspace english location flash:bacdprompts/
 param aa-pilot 4110
 param second-greeting-time 60
 param welcome-prompt _bacd_welcome.au
 param queue-manager-debugs 1000
 param call-retry-timer 15
 ?!
 service app-b-acd
 param queue-len 15
 param number-of-hunt-grps 1
 param aa-hunt2 
 param queue-manager-debugs 1
 ?!
 !
 !
 dial-peer voice 222 voip
 ?service app-b-acd-aa
 ?destination-pattern 4110
 ?session target ipv4:10.10.110.3
 ?incoming called-number 4110
 ?dtmf-relay h245-alphanumeric
 ?codec g711ulaw
 ?no vad
 !
 ___
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 visit www.ipexpert.com
 
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___
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Re: [OSL | CCIE_Voice] UCCX prompt recording

2012-01-22 Thread William Affeldt
Script

Create document variable
Just name the document. 

Accept step
Record.   Step  
Point record step at document variable
Save and upload to CCX 

After recording your prompt use windows search.

Search for *.wav 

order the files by date. The newest document will be your recording. Save to 
desktop and rename. 


Hopefully that helps. 

Sent from my iPhone

On Jan 22, 2012, at 11:37 AM, Randall Crumm rrcr...@yahoo.com wrote:

 HI,
 How are people recording custom prompts?
 
 Thanks
  
  Randall
 ___
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Re: [OSL | CCIE_Voice] UCCX prompt recording

2012-01-22 Thread William Affeldt
You actually don't need the prompt step. I use my script every lab and it still 
creates the prompt. 

Sent from my iPhone

On Jan 22, 2012, at 12:03 PM, William Affeldt william.affe...@yahoo.com wrote:

 Script
 
 Create document variable
 Just name the document. 
 
 Accept step
 Record.   Step  
 Point record step at document variable
 Save and upload to CCX 
 
 After recording your prompt use windows search.
 
 Search for *.wav 
 
 order the files by date. The newest document will be your recording. Save to 
 desktop and rename. 
 
 
 Hopefully that helps. 
 
 Sent from my iPhone
 
 On Jan 22, 2012, at 11:37 AM, Randall Crumm rrcr...@yahoo.com wrote:
 
 HI,
 How are people recording custom prompts?
 
 Thanks
  
  Randall
 ___
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 ___
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___
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Re: [OSL | CCIE_Voice] BR2-RTR-ntp-stratum config

2012-01-22 Thread William Affeldt
NTP takes a long time. I would do other things and check it in 20 min

Sent from my iPhone

On Jan 22, 2012, at 5:56 PM, Errol Abrahams eabraham2...@gmail.com wrote:

 
 
 Hi All,
 
 I have a problem with the ntp stratum config on BR2-RTR router as per the 
 config. I would like BR2-RTR to synch its ntp stratum clock with the 
 PSTN-Router LO0 address which is 10.10.100.2
 
 The address 10.10.100.2 is pingable from BR2-RTR and I have attached the 
 BR2-RTR config for your attention. I have also attached the message that I 
 get on BR2-RTR when I do 'show ntp stat' on this router.
 
 Thank you for your help.
 
 Cheers
 
 EA
 BR2-RTR--SH-NTP-STAT.txt
 BR2-RTR-Vol-Lab1A.txt
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Re: [OSL | CCIE_Voice] lab 6 released ):

2012-01-21 Thread William Affeldt
I would guess it is a variation of the bastard. I would not worry so much and 
just start saving for the next attempt. You will eventually get something that 
you can pass. 

Sent from my iPhone

On Jan 21, 2012, at 11:17 AM, sam max sammaxc...@gmail.com wrote:

 Hi Guys,
  
 I am so so so upsad this time was my last i believe but. cant blame 
 anyone except myself!!
  
 I got new lab 6 in brusels it was all changed early offer, MGCP TS, ipcc 
 changed, i am nt sure how we can complete it in 8 hrs time.
  
 I am really upsad now no money for next attempt nt sure when i can go back 
 for this,
  
 All the questions are comming in my eyes again and again but i was helpless
  
 Best of luck guys who are going next ...
  
 Regards
 Sam
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Re: [OSL | CCIE_Voice] Type and Plan on H323

2012-01-21 Thread William Affeldt
The easiest way is to use a isdn map. You can always use translation rule and 
profile too.  I would not set anything in CCM for h323. 

Sent from my iPhone

On Jan 21, 2012, at 12:09 PM, datucha123 datucha123 datucha...@gmail.com 
wrote:

 Hello,
 
 H323 Router by default (in IOS) sets the Calling Type and Plan for outgoing 
 calls to PRI. - I think there is something put in IOS so that it recognizes 
 the US dialplan, and set the Type and Plan automatically.
 
 How can I disable it?
 
 Because when I set the Type and Plan in Callmanager, and send the call to 
 h323 gateway, if the Calling Number matches the US dialplan, then the H323 
 gateway overwrites the Type and Plan.
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[OSL | CCIE_Voice] Issues connecting to proctorlabs?

2011-12-10 Thread William Affeldt
Anyone else having issues? Website is down and VPN.___
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[OSL | CCIE_Voice] Silent Moh File

2011-12-08 Thread William Affeldt
Does anyone have a silent Moh file already created that I can use?___
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Re: [OSL | CCIE_Voice] Silent Moh File

2011-12-08 Thread William Affeldt
That is a great Idea. 

Sent from my iPhone

On Dec 8, 2011, at 11:57 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote:

 You can get it from the UCCX server in the prompt file  ,
 
 Ash
 
 
 
 On Thu, Dec 8, 2011 at 1:25 PM, William Affeldt
 william.affe...@yahoo.com wrote:
 Does anyone have a silent Moh file already created that I can use?
 
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Re: [OSL | CCIE_Voice] Problem with Multicast MOH from router flash

2011-11-30 Thread William Affeldt
Make sure that you configured the codec under the dial peers on BR1. They use 
g729 by default. Sounds like the dial-peers to call manager from BR1 don't have 
a voice class codec or are a hard set codec configured. 

Sent from my iPhone

On Nov 30, 2011, at 8:12 AM, Rynard Coetzee rynard.coet...@bytes.co.za wrote:

 Hi guys
 
 I have been struggling with this problem for most of the day now ,and I don`t 
 know what else I can check. The MoH works if I call from HQ branch to BR1 
 ,but when I call from the PSTN to BR1 I get tone on hold. I have the relevant 
 config on my BR1 router
 
 Ccm-manager music-on-hold
 
 Telephony service
 
 Moh music-on-hold.au
 
 Multicast moh 239.1.1.1 port 16384 route 10.10.110.2 (loopback) 10.10.201.1 
 (Voice Int)
 
 I also configured separate DP for MOH server in CUCM with a G711 only region. 
 I have the MOH setting on the audio source ,server and MRG set. Another 
 strange thing that I see is when I call from HQ phone to BR1 phone and I put 
 BR1 phone on hold I hear the MOH stream ,but when I do show ccm-manager 
 music-on-hold it shows that there are 0 active multicast sessions ?
 
 Any ideas ,because I have run out … ?
 Regards
 
 Rynard
 
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[OSL | CCIE_Voice] Proctorlabs down?

2011-11-28 Thread William Affeldt
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Re: [OSL | CCIE_Voice] lab - 6

2011-11-21 Thread William Affeldt
I am 99.9 percent sure that the version in the lab only supports Software MTP 
at g711. I am sure Vic can confirm.

Sent from my iPhone

On Nov 21, 2011, at 1:33 PM, Bartosz Sokolowski ibartosz.sokolow...@gmail.com 
wrote:

 You are wrong.
 Early Offer requires MTP. CCM doesn't care how you provide MTP. 
 It's possible to configure software MTP in IOS with G711 or G729. If we 
 consider software MTP it's G711 or (not and!) G729. If you use hardware MTP 
 on IOS it's G711 and/or G729.
 -- 
 Regards,
 Bartosz
 
 2011/11/21 Peter Jeff peterjeff2...@yahoo.com
 But early offer required MTP hardware if i am nt wrong
 
 From: Mohamed Hassan mrmha...@gmail.com
 To: Abel ... midga...@gmail.com 
 Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com 
 Sent: Monday, November 21, 2011 12:28 PM
 
 Subject: Re: [OSL | CCIE_Voice] lab - 6
 
 SIP Early offer it is IOS command, it is equal to the configuration of fast 
 start on CUCM
 
 http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-sipsip.html#wp1351645
  
 
 It can be configured globally
 SUMMARY STEPS
 
 1. enable
 2. configure terminal
 3. voice service voip
 4. allow-connections sip
 5. early-offer forced
 6. exit
 
 
 and it can be configured under dial-peer
 SUMMARY STEPS
 
 1. enable
 2. configure terminal
 3. dial-peer voice 1 voip
 4. voice-class sip early-offer forced
 5. exit
 
 
 On Mon, Nov 21, 2011 at 2:48 AM, Abel ... midga...@gmail.com wrote:
 Well, the DHCP issue can be done easy if you look 5 minutes on cisco.com 
 documentation, by Early offer what you mean? no interdigit timeout?
 
 
 On Sun, Nov 20, 2011 at 2:22 PM, datucha123 datucha123 datucha...@gmail.com 
 wrote:
 Do you mean under the DHCP configuration, that such question has been on the 
 exam?
  
 What do you mean under the SIP early offer? or MGCP TS?
  
 
 
  
 On Sun, Nov 20, 2011 at 8:24 PM, Peter Jeff peterjeff2...@yahoo.com wrote:
 Hi Guys,
 
 It was my 5th attempt and i got lab 6 frustration is on peak everytime i went 
 for the lab i get new lab
 
 From last 2 months all my JR guys passed in dubai bec they got lab 3 and lab 
 4 in Dubai since last 2 months i saw lot of guys cleared from dubai 
 
 Now lab 6 i am so so so frustrated its all new 
 
 Configure the local Cisco router R3 as DHCP server to provide ip add  for SC 
 from local subnet , MAke sure SCPH1 get ip add 142.102.66.11 and SCPH2 shld 
 be 142.102.66.12 YOU ARE ALLOWED TO CREATE ONE DHCP POOL FOR ALL SC PHONES
 
 SIP early offer , MGCP TS which some different issue.
 
 best of luck guys my Christmas will go bad this time as no dates available in 
 Dubai now. Its my bad sunday
 
 Regards
 Jeff
  
 
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 ___
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 ___
 For more information regarding industry leading CCIE Lab training, please 
 visit www.ipexpert.com
 
 Are you a CCNP or CCIE and looking for a job? Check out 
 www.PlatinumPlacement.com
 
 
 
 -- 
 Engineer / Mohamed Rabea
 Unified communication engineer
 
 ___
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 ___
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 ___
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Re: [OSL | CCIE_Voice] lab - 6

2011-11-21 Thread William Affeldt
The question probably does not restrict you to one DHCP pool.

Sent from my iPhone

On Nov 20, 2011, at 11:41 PM, Ken Wyan kew...@gmail.com wrote:

 You can't provide 2 static ip addresses using a single dhcp pool on a cisco 
 router
  
 Either you have misunderstood or a typo in exam.
 
 
  
 On Mon, Nov 21, 2011 at 5:18 AM, Abel ... midga...@gmail.com wrote:
 Well, the DHCP issue can be done easy if you look 5 minutes on cisco.com 
 documentation, by Early offer what you mean? no interdigit timeout?
 
 
 On Sun, Nov 20, 2011 at 2:22 PM, datucha123 datucha123 datucha...@gmail.com 
 wrote:
 Do you mean under the DHCP configuration, that such question has been on the 
 exam?
  
 What do you mean under the SIP early offer? or MGCP TS?
  
 
 
  
 On Sun, Nov 20, 2011 at 8:24 PM, Peter Jeff peterjeff2...@yahoo.com wrote:
 Hi Guys,
 
 It was my 5th attempt and i got lab 6 frustration is on peak everytime i went 
 for the lab i get new lab
 
 From last 2 months all my JR guys passed in dubai bec they got lab 3 and lab 
 4 in Dubai since last 2 months i saw lot of guys cleared from dubai 
 
 Now lab 6 i am so so so frustrated its all new 
 
 Configure the local Cisco router R3 as DHCP server to provide ip add  for SC 
 from local subnet , MAke sure SCPH1 get ip add 142.102.66.11 and SCPH2 shld 
 be 142.102.66.12 YOU ARE ALLOWED TO CREATE ONE DHCP POOL FOR ALL SC PHONES
 
 SIP early offer , MGCP TS which some different issue.
 
 best of luck guys my Christmas will go bad this time as no dates available in 
 Dubai now. Its my bad sunday
 
 Regards
 Jeff
  
 
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Re: [OSL | CCIE_Voice] Why my outbound calls (using RGs) to pstn doesn't work?

2011-11-15 Thread William Affeldt
Check that your h323 bind interface is up and reachable from the phone and call 
manager. Also reset the gateway. Finally, if everything is up and pingable do a 
debug voice ccapi inout to verify the matched dial peers. It sounds to me like 
you bound the vlan interface and it is down because of a inactive vlan or no 
ports are up and assigned to the VLAN. 

Sent from my iPhone

On Nov 15, 2011, at 11:42 AM, Voiper datapack...@gmail.com wrote:

 Unfortunately, I am still stuck. I took two days off from work (Mon, Tue) to 
 work on it and still  no success 33D.gif
 Will keep trying.
 
 Thanks,
 
 Voiper
 
 On Tue, Nov 15, 2011 at 2:35 PM, Rrcrumm rrcr...@yahoo.com wrote:
 We're u able to resolve your issue?
 
 Sent from my iPhone
 
 On Nov 15, 2011, at 10:32 AM, Voiper datapack...@gmail.com wrote:
 
 Hi Friends:
 
 Thank you all for you quick and helpful suggestions. It show you are all 
 very focused and have the determination to get the coveted number. I need to 
 step up my efforts after seeing how you all  quickly helped with feedback. 
 
 All your help is truly appreciated.
 
 Voiper
 
 
 @RRCRUMM:
 Did you reset the h323 gw in cucm? Do you see an ip address towards the 
 upper portion of the page above the name/ ip address field?
 - The h323 gw is configured and I see the ip address
 outbound calls to pstn using h323 or mcgp gw directly is working
 
 @Ashwani ash_r...@hotmail.com 
 Did you bind source interface on HQ router for H323? And same interface's IP 
 assigned on CUCM?  Also did you have incoming dial-peer from CUCM? Paste 
 your HQ config here.
 - interface FastEthernet0/0.20
  description VOICE
  encapsulation dot1Q 20
  ip address 10.10.200.3 255.255.255.0
  ip helper-address 10.10.210.10
  h323-gateway voip bind srcaddr 10.10.200.3
 
 HQ router config is attached.
 
 @Ken Wyan kew...@gmail.com
 Pls check your route list  rl-test config  digit manipulations etc.. in 
 rg-test (under rl-test).
 If you have done any digit manipulations in rg-test (under rl-test) , your 
 RP digit-manipulations will not work
 Add rg-test  as Local Route Group for Device Pool
 In addition to debug isdn ---  debug ccapi inout , dialpeer  may help to 
 isolate issue
 - No digit manipulations on any of the above. I tried to keep it as simple 
 as possible
 ran above debug and nothing gets generated. 
 
 @Pradeep Kumar Sharma sonu.netwo...@gmail.com
 Verify the registration status of your Route List.
 Try to reset it once.
 Run the debug voice ccapi inout on H323 Gw and see if the outbound calls 
 really hitting to the gateway on not ?
 - route list status unknown. Reset many times. What is your registration 
 status?
 
 CCAPI:  debug voip ccapi inout is ON (filter is OFF)
 CCH323 SPI: H225 State Machine tracing is enabled (filter is OFF)
 - above debug show nothing happening. Call just not reaching the h323 gw
 
 
 @Mohd Baqari baqari.voic...@gmail.com 
 On you gateway, do the following:
 1. Debug cch323 h225 messages. Check if any h323 signaling is reaching the 
 gateway.
 2. If signaling is received successfully, run the command debug voice 
 dialpeer. Check which dial peer is matched and whats the configuration of it
 3. If no h323 signaling is received, post cucm traces after making a call
 - above debug show nothing happening. Call just not reaching the h323 gw
 Do I need to run RTMT for trace? I installed it on my desktop and ran trace 
 'Collect Files' (attached). First time using, so I don't know if I got the 
 right one?
 
 
 ccm0007.txt
 ccm0007.txt
 SDL002_100_10.txt
 HQ-RTR.txt
 
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Re: [OSL | CCIE_Voice] Quick Question

2011-11-01 Thread William Affeldt
If by wall jack you mean analog line from the provider then just plug it into 
one of the FXO lines and PLAR the call to either the auto attendant or a phone. 
Outbound calling will work also, just not at the same time. 

Sent from my iPhone

On Nov 1, 2011, at 7:17 PM, Jonathan Preston jonathan_preston2...@yahoo.com 
wrote:

 I have a quick question.  I have to do a demo with a UC500 at a hotel and the 
 onsite engineer will only allow a connections into a wall Jack.  My question 
 is can I connect the FXS port to the wall jack and route a call from the 
 PSTN.  The hotel is going to provide a DID for the demo.  They will not let 
 us connect directly into switch port on their PBX.  He said he would provide 
 me an IP Address but no connection to the PBX.
 
 ___
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 visit www.ipexpert.com
 
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 www.PlatinumPlacement.com
___
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Re: [OSL | CCIE_Voice] Downloading List.xml from Call Manager.

2011-10-24 Thread William Affeldt
Hi All, 
 
Thank you for the replys. 
 
The error from RTMT is File[List.xml] not found. Also, all of you are referring 
to sending the file to the phone. I am trying to download the file from call 
manager to BR2. I know the error code seems strait forward but it is not. The 
file List.xml is in the root directory. I have also uploaded three other files 
into the same / directory and they work fine. I have deleted it and uploaded it 
again. I have had two other sets of eyes look at the spelling and directory 
placement. Can someone else try to upload the List.xml file into the root 
directory and download it to BR2. 
 



From: Chris Martin clm.c...@gmail.com
To: William Affeldt william.affe...@yahoo.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Monday, October 24, 2011 1:51 PM
Subject: Re: [OSL | CCIE_Voice] Downloading List.xml from Call Manager.


Just to add, remember everything is case sensitive.. I have shot myself in the 
foot more than once when uploading files, both the name and the directory 
structure.  IE: Desktops/320x212x12 does not equal desktops/320x312x12..  If 
placed in the wrong spot the phone will not find it.  You can see what is being 
requested in UCM traces.

Chris


On Mon, Oct 24, 2011 at 12:32 PM, William Affeldt william.affe...@yahoo.com 
wrote:





Ok, so I have to be missing something. I can't download the List.xml file from 
Call Manager but I can download my image files. I have triple checked the file 
name and the directory. I am downloading the file from same directory as the 
images and they work fine. Is there something I am missing to download the 
file? Also, I downloaded different .xml file just fine.
___
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visit www.ipexpert.com

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www.PlatinumPlacement.com
___
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Re: [OSL | CCIE_Voice] Downloading List.xml from Call Manager.

2011-10-24 Thread William Affeldt
It is going to root because I am trying to transfer the file from call manager 
to the BR2-RTR not a phone. 

Sent from my iPhone

On Oct 24, 2011, at 2:33 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:

 unfortunately i dont have lab right now that's why my answers is weak
 because i relay on my memory when i studied this feature ,
 
 but why you are uploading the List.xml to the root ? it should place
 on the same directory as the Photos
 
 Ash
 
 On Mon, Oct 24, 2011 at 4:12 PM, William Affeldt
 william.affe...@yahoo.com wrote:
 Hi All,
 
 Thank you for the replys.
 
 The error from RTMT is File[List.xml] not found. Also, all of you are
 referring to sending the file to the phone. I am trying to download the file
 from call manager to BR2. I know the error code seems strait forward but it
 is not. The file List.xml is in the root directory. I have also uploaded
 three other files into the same / directory and they work fine. I have
 deleted it and uploaded it again. I have had two other sets of eyes look at
 the spelling and directory placement. Can someone else try to upload the
 List.xml file into the root directory and download it to BR2.
 
 From: Chris Martin clm.c...@gmail.com
 To: William Affeldt william.affe...@yahoo.com
 Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Sent: Monday, October 24, 2011 1:51 PM
 Subject: Re: [OSL | CCIE_Voice] Downloading List.xml from Call Manager.
 
 Just to add, remember everything is case sensitive.. I have shot myself in
 the foot more than once when uploading files, both the name and the
 directory structure.  IE: Desktops/320x212x12 does not equal
 desktops/320x312x12..  If placed in the wrong spot the phone will not find
 it.  You can see what is being requested in UCM traces.
 
 Chris
 
 On Mon, Oct 24, 2011 at 12:32 PM, William Affeldt
 william.affe...@yahoo.com wrote:
 
 
 Ok, so I have to be missing something. I can't download the List.xml file
 from Call Manager but I can download my image files. I have triple checked
 the file name and the directory. I am downloading the file from same
 directory as the images and they work fine. Is there something I am missing
 to download the file? Also, I downloaded different .xml file just fine.
 ___
 For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com
 
 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com
 
 
 
 
 ___
 For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com
 
 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com
 
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com


Re: [OSL | CCIE_Voice] Downloading List.xml from Call Manager.

2011-10-24 Thread William Affeldt
I don't think so but, every other file was working fine. I have a lab in the 
morning and I will try restarting the service and see if that corrects the 
issue. 

Sent from my iPhone

On Oct 24, 2011, at 3:00 PM, clm.c...@gmail.com wrote:

 Assume you restarted the tftp services after uploading?
 
 Chris
 Sent on the Sprint® Now Network from my BlackBerry®
 
 -Original Message-
 From: William Affeldt william.affe...@yahoo.com
 Date: Mon, 24 Oct 2011 14:49:37 
 To: Ashraf Ayyashash.ayy...@gmail.com
 Cc: Chris Martinclm.c...@gmail.com; 
 ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] Downloading List.xml from Call Manager.
 
 It is going to root because I am trying to transfer the file from call 
 manager to the BR2-RTR not a phone. 
 
 Sent from my iPhone
 
 On Oct 24, 2011, at 2:33 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
 
 unfortunately i dont have lab right now that's why my answers is weak
 because i relay on my memory when i studied this feature ,
 
 but why you are uploading the List.xml to the root ? it should place
 on the same directory as the Photos
 
 Ash
 
 On Mon, Oct 24, 2011 at 4:12 PM, William Affeldt
 william.affe...@yahoo.com wrote:
 Hi All,
 
 Thank you for the replys.
 
 The error from RTMT is File[List.xml] not found. Also, all of you are
 referring to sending the file to the phone. I am trying to download the file
 from call manager to BR2. I know the error code seems strait forward but it
 is not. The file List.xml is in the root directory. I have also uploaded
 three other files into the same / directory and they work fine. I have
 deleted it and uploaded it again. I have had two other sets of eyes look at
 the spelling and directory placement. Can someone else try to upload the
 List.xml file into the root directory and download it to BR2.
 
 From: Chris Martin clm.c...@gmail.com
 To: William Affeldt william.affe...@yahoo.com
 Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Sent: Monday, October 24, 2011 1:51 PM
 Subject: Re: [OSL | CCIE_Voice] Downloading List.xml from Call Manager.
 
 Just to add, remember everything is case sensitive.. I have shot myself in
 the foot more than once when uploading files, both the name and the
 directory structure.  IE: Desktops/320x212x12 does not equal
 desktops/320x312x12..  If placed in the wrong spot the phone will not find
 it.  You can see what is being requested in UCM traces.
 
 Chris
 
 On Mon, Oct 24, 2011 at 12:32 PM, William Affeldt
 william.affe...@yahoo.com wrote:
 
 
 Ok, so I have to be missing something. I can't download the List.xml file
 from Call Manager but I can download my image files. I have triple checked
 the file name and the directory. I am downloading the file from same
 directory as the images and they work fine. Is there something I am missing
 to download the file? Also, I downloaded different .xml file just fine.
 ___
 For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com
 
 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com
 
 
 
 
 ___
 For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com
 
 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com
 
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] Downloading List.xml from Call Manager.

2011-10-24 Thread William Affeldt
Thank you sir. I forget to do that every time with TFTP. 

Sent from my iPhone

On Oct 24, 2011, at 3:18 PM, Chris Martin clm.c...@gmail.com wrote:

 Just did a quick test...
 
 Uploaded List.xml to root of my test Pub.
 
 BR1-2851#copy tftp flash
 Address or name of remote host []? 198.183.241.210
 Source filename []? List.xml
 Destination filename [List.xml]?
 Accessing tftp://198.183.241.210/List.xml...
 %Error opening tftp://198.183.241.210/List.xml (No such file or directory)
 BR1-2851#
 
 --
 
 ssh to PUB:
 
 admin:file list activelog /cm/trace/ctftp/sdi/
 
 ctftp0001.txt   ctftp0002.txt
 ctftp0003.txt   ctftp0004.txt
 ctftp0005.txt   ctftp0006.txt
 ctftp0007.txt   ctftp0008.txt
 ctftp0009.txt   ctftp~num.bin
 dir count = 0, file count = 10
 
 admin:file tail activelog /cm/trace/ctftp/sdi/ctftp0009.txt
 
 10/24/2011 19:13:28.105 TFTP|   
 CReqContext::tftp[0x9223858~0~10.101.70.20~59076] File[List.xml] not 
 found|CLID::StandAloneClusterNID::CM-PUBLVL::ErrorMASK::0004
 
 ---
 
 Restarted TFTP services:
 
 BR1-2851#copy tftp flash
 Address or name of remote host [198.183.241.210]?
 Source filename [List.xml]?
 Destination filename [List.xml]?
 Accessing tftp://198.183.241.210/List.xml...
 Loading List.xml from 198.183.241.210 (via GigabitEthernet0/0): !
 [OK - 8 bytes]
 
 8 bytes copied in 0.184 secs (43 bytes/sec)
 BR1-2851#
 
 Successful.
 
 On Mon, Oct 24, 2011 at 5:08 PM, William Affeldt william.affe...@yahoo.com 
 wrote:
 I don't think so but, every other file was working fine. I have a lab in the 
 morning and I will try restarting the service and see if that corrects the 
 issue.
 
 Sent from my iPhone
 
 On Oct 24, 2011, at 3:00 PM, clm.c...@gmail.com wrote:
 
  Assume you restarted the tftp services after uploading?
 
  Chris
  Sent on the Sprint® Now Network from my BlackBerry®
 
  -Original Message-
  From: William Affeldt william.affe...@yahoo.com
  Date: Mon, 24 Oct 2011 14:49:37
  To: Ashraf Ayyashash.ayy...@gmail.com
  Cc: Chris Martinclm.c...@gmail.com; 
  ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com
  Subject: Re: [OSL | CCIE_Voice] Downloading List.xml from Call Manager.
 
  It is going to root because I am trying to transfer the file from call 
  manager to the BR2-RTR not a phone.
 
  Sent from my iPhone
 
  On Oct 24, 2011, at 2:33 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
 
  unfortunately i dont have lab right now that's why my answers is weak
  because i relay on my memory when i studied this feature ,
 
  but why you are uploading the List.xml to the root ? it should place
  on the same directory as the Photos
 
  Ash
 
  On Mon, Oct 24, 2011 at 4:12 PM, William Affeldt
  william.affe...@yahoo.com wrote:
  Hi All,
 
  Thank you for the replys.
 
  The error from RTMT is File[List.xml] not found. Also, all of you are
  referring to sending the file to the phone. I am trying to download the 
  file
  from call manager to BR2. I know the error code seems strait forward but 
  it
  is not. The file List.xml is in the root directory. I have also uploaded
  three other files into the same / directory and they work fine. I have
  deleted it and uploaded it again. I have had two other sets of eyes look 
  at
  the spelling and directory placement. Can someone else try to upload the
  List.xml file into the root directory and download it to BR2.
 
  From: Chris Martin clm.c...@gmail.com
  To: William Affeldt william.affe...@yahoo.com
  Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
  Sent: Monday, October 24, 2011 1:51 PM
  Subject: Re: [OSL | CCIE_Voice] Downloading List.xml from Call Manager.
 
  Just to add, remember everything is case sensitive.. I have shot myself in
  the foot more than once when uploading files, both the name and the
  directory structure.  IE: Desktops/320x212x12 does not equal
  desktops/320x312x12..  If placed in the wrong spot the phone will not find
  it.  You can see what is being requested in UCM traces.
 
  Chris
 
  On Mon, Oct 24, 2011 at 12:32 PM, William Affeldt
  william.affe...@yahoo.com wrote:
 
 
  Ok, so I have to be missing something. I can't download the List.xml file
  from Call Manager but I can download my image files. I have triple checked
  the file name and the directory. I am downloading the file from same
  directory as the images and they work fine. Is there something I am 
  missing
  to download the file? Also, I downloaded different .xml file just fine.
  ___
  For more information regarding industry leading CCIE Lab training, please
  visit www.ipexpert.com
 
  Are you a CCNP or CCIE and looking for a job? Check out
  www.PlatinumPlacement.com
 
 
 
 
  ___
  For more information regarding industry leading CCIE Lab training, please
  visit

Re: [OSL | CCIE_Voice] CCIE San Jose Exam Lab Equipment Problems

2011-10-13 Thread William Affeldt
Yes, I had my subscriber and cups lock up in the middle of the lab. The proctor 
had to reload my whole pod server wise. Also, during the same exam my cue was 
not reachable randomly. I had to reload it from the cue multiple times during 
the lab to restore ip connectivity. Since It looked like I got all of unity 
connection right and zero in cue based on my score report. I am assuming it was 
not working during grading.

Sent from my iPhone

On Oct 12, 2011, at 10:15 PM, Erwan Erwan e_er...@yahoo.com wrote:

 hi all,
  
 Does anyone had experienced  Voice Lab equipment issue/problems in San Jose 
 testing center?
  
 Could share the experience here  ?
  
  
 I see the old posting saying someone experienced the HW issue during the lab 
 exam (below)
 
 Norm Hinman
 Wed, 07 Jan 2009 16:53:46 -0800
 Next time, if this happens and they won't swap the PC, I am demanding to 
 speak 
 to a manager.  The more I think about it, the angrier I get.  that was MY 
 $1,400!
 
 What ever happened to the concept of taking care of the customer?
   - Original Message - 
   From: Jimmy Lister 
   To: Norm Hinman ; ccie_secur...@onlinestudylist.com 
   Sent: Tuesday, January 06, 2009 9:26 PM
   Subject: RE: [OSL | CCIE_Security] CCIE San Jose Exam Lab Equipment Problems
 
 
   Norm,
 
   Thanks for the heads up.  I am scheduled for San Jose next Friday; hope I 
 don't get that workstation!  I just checked and there are a lot of dates open 
 in Feb at San Jose but RTP is full until April.  There must be limited seats 
 for security at RTP or people just prefer RTP. 
 
 
 Thanks
 ___
 For more information regarding industry leading CCIE Lab training, please 
 visit www.ipexpert.com
 
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 www.PlatinumPlacement.com

___
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Re: [OSL | CCIE_Voice] Has anyone been able to connect to his servers today (hardware vpn)

2011-10-03 Thread William Affeldt
My VPN is up this morning but my servers are down. 

Sent from my iPhone

On Oct 3, 2011, at 5:14 AM, pithog...@yahoo.com wrote:

 Hi experts,
 
 Has any one been able to connect to his servers on proctorlabs today? I 
 struggled to connect to my servers for more than 5 hours but I couldn't all I 
 could do was to ping the servers from my router.
 
 Please assist if you have the fix.
 
 Regards
 
 Sola Ayoade
 Sent from my BlackBerry® Smartphone, from Etisalat. Enjoy high speed internet 
 service with Etisalat easy net, available at all our experience centres
 ___
 For more information regarding industry leading CCIE Lab training, please 
 visit www.ipexpert.com
 
 Are you a CCNP or CCIE and looking for a job? Check out 
 www.PlatinumPlacement.com
___
For more information regarding industry leading CCIE Lab training, please visit 
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Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] I need hardware Vpn assistance on a session now

2011-10-02 Thread William Affeldt
VPN is down for me and a couple of other people I have talked to. 

Sent from my iPhone

On Oct 2, 2011, at 7:26 AM, Randall.Crumm randall.cr...@gmail.com wrote:

 If you can't reach the server web GUI via IP, your issue is infrastructure.
 Check the config for hq rtr and your 3750
 Hth
 Randall
 
 Sent from my iPhone
 
 On Oct 1, 2011, at 11:25 PM, pithog...@yahoo.com wrote:
 
 Experts thanks for your assistance.
 
 Attached to this mail is my config , I can access my Pod but the challenge 
 is that the Pc I use for the session should connect to fa 3 according to my 
 config and also give me access to my servers GUI is not working for me, 
 connection to my pod was via console, now I want to do away with console and 
 connect to int fa 3 but my laptop says dns unresolved.
 
 Please take a look at my config and alert me if you have a work through.
 
 Regards
 
 Pithog oil
 Sent from my BlackBerry® Smartphone, from Etisalat. Enjoy high speed 
 internet service with Etisalat easy net, available at all our experience 
 centres
 
 -Original Message-
 From: Rrcrumm rrcr...@yahoo.com
 Date: Sat, 1 Oct 2011 21:47:39 
 To: pithog...@yahoo.compithog...@yahoo.com
 Cc: Marko Milivojevicmar...@ipexpert.com; edgar 
 felizejfeli...@yahoo.com; 
 ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com; 
 supp...@ipexpert.comsupp...@ipexpert.com; 
 supp...@proctorlabs.comsupp...@proctorlabs.com
 Subject: Re: [OSL | CCIE_Voice] I need hardware Vpn assistance on a session 
 now
 
 Are you consoled onto the router? You need to do a command from the CLI to 
 VPN in. O can't remember it offhand but it us something like, IPSec VPN 
 client ezvpn connect, then the Cauthen command. Below you said you computer 
 can't reach the Internet. You should use the router CLI.
 
 Hth
 Randall
 
 Sent from my iPhone
 
 On Oct 1, 2011, at 2:00 PM, pithog...@yahoo.com wrote:
 
 The support people at ipexpert have not been helpful in recent times all 
 they tell me is that there are not ccie level support engineers, I just 
 changed my hardware vpn to a 881 router last week so its been difficult to 
 ascertain where the issue is coming from although I have done all I should 
 do on my end and I had issues through out this morning, as I speak I have 
 just started a new session, I will appreciate some support now.
 
 Thanks marko
 Sent from my BlackBerry® Smartphone, from Etisalat. Enjoy high speed 
 internet service with Etisalat easy net, available at all our experience 
 centres
 
 -Original Message-
 From: Marko Milivojevic mar...@ipexpert.com
 Date: Sat, 1 Oct 2011 13:51:57 
 To: edgar felizejfeli...@yahoo.com
 Cc: Chris Martinclm.c...@gmail.com; 
 pithog...@yahoo.compithog...@yahoo.com; 
 ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] I need hardware Vpn assistance on a session 
 now
 
 We've had somewhat a perfect storm of multiple things go wrong. We
 should be operational right now and we have several support cases open
 with our ISPs and equipment vendors.
 
 Please, get in touch with our support at supp...@ipexpert.com and
 supp...@proctorlabs.com with any issues that you may still be having.
 We have our entire support and technical team, including all the
 instructors working on maintaing our services up and running at the
 level you are all used to.
 
 We sincerely apologize for the inconveniences.
 
 --
 Marko Milivojevic - CCIE #18427
 Senior Technical Instructor - IPexpert
 
 FREE CCIE training: http://bit.ly/vLecture
 
 Mailto: mar...@ipexpert.com
 Telephone: +1.810.326.1444
 Web: http://www.ipexpert.com/
 
 On Sat, Oct 1, 2011 at 11:34, edgar feliz ejfeli...@yahoo.com wrote:
 To many people are having issue for it to be operator error
 it's definitely something on their end ISP etc.
 EJF
 
 From: Chris Martin clm.c...@gmail.com
 To: pithog...@yahoo.com
 Cc: ccie_voice@onlinestudylist.com
 Sent: Saturday, October 1, 2011 10:31 AM
 Subject: Re: [OSL | CCIE_Voice] I need hardware Vpn assistance on a session
 now
 
 You can open a case with support, that would be the quickest way to get
 their help.  Did you use IPE's configuration for hardware vpn?  What type 
 of
 router are you using?  Did you enter the commands to initiate the xauth?
 Chris
 
 On Sat, Oct 1, 2011 at 5:58 AM, pithog...@yahoo.com wrote:
 
 I am on a current session now and I can't access proctorlabs via hardware
 vpn, my router can ping internet but my laptop connected to the router 
 can't
 access the internet so as to initiate hardware VPN .
 
 Please help, you can even connect to my laptop via team viewer .
 
 Experts kindly assist
 
 Pithohoil
 Sent from my BlackBerry® Smartphone, from Etisalat. Enjoy high speed
 internet service with Etisalat easy net, available at all our experience
 centres
 ___
 For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com
 
 Are you a CCNP or 

Re: [OSL | CCIE_Voice] Proctor Labs down again?

2011-10-02 Thread William Affeldt
I am still down. 

Sent from my iPhone

On Oct 2, 2011, at 10:44 AM, Brian Mulgrew btmulg...@hotmail.com wrote:

 Hi - has anyone been able to connect to the voice racks since yesterday's 
 outage?
  
 I have logged a case but was advised 5 hours ago that there is still a vpn 
 issue with no eta for a fix.
  
 Cheers
 Brian
 ___
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 visit www.ipexpert.com
 
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 www.PlatinumPlacement.com
___
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Re: [OSL | CCIE_Voice] Proctor Labs site

2011-10-01 Thread William Affeldt
Yes, it appears to be down for me too. 

Sent from my iPhone

On Oct 1, 2011, at 5:33 AM, Larry Stern larry.st...@blackbox.com wrote:

 Before I contact proctor labs tech support, is anyone else having extremely 
 slow service on proctor labs site. Possibly all the racks are in use after 
 being blocked out during the two week boot camp period and everyone is 
 logging on at once? But I never saw it this slow.
 
 Sent from my iPad
 
 On Oct 1, 2011, at 5:15 AM, ccie_voice-requ...@onlinestudylist.com 
 ccie_voice-requ...@onlinestudylist.com wrote:
 
 Send CCIE_Voice mailing list submissions to
   ccie_voice@onlinestudylist.com
 
 To subscribe or unsubscribe via the World Wide Web, visit
   http://onlinestudylist.com/mailman/listinfo/ccie_voice
 or, via email, send a message with subject or body 'help' to
   ccie_voice-requ...@onlinestudylist.com
 
 You can reach the person managing the list at
   ccie_voice-ow...@onlinestudylist.com
 
 When replying, please edit your Subject line so it is more specific
 than Re: Contents of CCIE_Voice digest...
 
 
 Today's Topics:
 
  1. Re: MRGMRGL (Matthew Saskin)
  2. Re: MRGMRGL (Mohd Baqari)
  3. Re: MOH with H323 gateway (Vega Wong)
 
 
 --
 
 Message: 1
 Date: Fri, 30 Sep 2011 12:00:27 -0500
 From: Matthew Saskin m...@saskin.net
 To: eng_firasoq...@yahoo.com eng_firasoq...@yahoo.com
 Cc: ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] MRGMRGL
 Message-ID:
   camsv-mt-qddjq0q-jujh2w8zq9-y8ispwzqmqhm-dumf1cx...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1
 
 It shouldn't matter provided, in the example provided, each MRG contains
 only resources of the specified type.  Precedence for a MRG within an MRGL
 should only come into play if your MRG's contain the same types of
 resources.
 
 On Fri, Sep 30, 2011 at 8:14 AM, eng_firasoq...@yahoo.com 
 eng_firasoq...@yahoo.com wrote:
 
 Dear All,
 
 If we have to configure MTP,Xcoder,hd_Conference in the HQ site then we
 will configure 3 MRG MTP_MRG,Xcoder_MRG,conf_MRG after that we will create
 MRGL for hd_hq_MRGL but now what is correct order for MRG inside the MRGL
 
 Regards
 
 
 
 ___
 For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com
 
 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com
 
 -- next part --
 An HTML attachment was scrubbed...
 URL: 
 /archives/ccie_voice/attachments/20110930/9ad04614/attachment-0001.html
 
 --
 
 Message: 2
 Date: Fri, 30 Sep 2011 21:35:30 +0400
 From: Mohd Baqari baqari.voic...@gmail.com
 To: eng_firasoq...@yahoo.com eng_firasoq...@yahoo.com
 Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] MRGMRGL
 Message-ID: 77e4a2f2-eeae-491c-a441-c0f073fbe...@gmail.com
 Content-Type: text/plain; charset=us-ascii
 
 No need for 3 mrgs. Create one mrg and put all 3 resources into it. Order 
 doesnt matter since those resources are of different types.
 
 Regards,
 Mohammed Al Baqari
 
 Sent from my iPhone
 
 On Sep 30, 2011, at 5:14 PM, eng_firasoq...@yahoo.com 
 eng_firasoq...@yahoo.com wrote:
 
 Dear All,
 
 If we have to configure MTP,Xcoder,hd_Conference in the HQ site then we 
 will configure 3 MRG MTP_MRG,Xcoder_MRG,conf_MRG after that we will create 
 MRGL for hd_hq_MRGL but now what is correct order for MRG inside the MRGL
 
 Regards
 
 
 ___
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 visit www.ipexpert.com
 
 Are you a CCNP or CCIE and looking for a job? Check out 
 www.PlatinumPlacement.com
 -- next part --
 An HTML attachment was scrubbed...
 URL: 
 /archives/ccie_voice/attachments/20110930/99ab822b/attachment-0001.html
 
 --
 
 Message: 3
 Date: Sat, 1 Oct 2011 17:16:39 +1000
 From: Vega Wong vega2...@yahoo.com.au
 To: ccie_voice@onlinestudylist.com
 Subject: Re: [OSL | CCIE_Voice] MOH with H323 gateway
 Message-ID: 01cc800a$109d3bf0$31d7b3d0$@com.au
 Content-Type: text/plain; charset=us-ascii
 
 Hi
 
 
 
 I have investigated further into this issue, and I think I may have found
 the reason for this.
 
 As the MOH server is setup with 1 hop limit with the multicast setting, and
 the HQ router/GW is only 1 hop away, so the moh stream from CUCM can reach
 to the loopback address of the HQ router within 1 hop. But correct me if I
 am wrong, that could be the reason why the moh stream from CUCM can reach
 PSTN caller.
 
 
 
 So my question is, is there a way to make the HQ loopback more than 1 hop
 away?
 
 
 
 Thanks
 
 
 
 
 
 
 
 
 
 
 
 From: Bill Lake [mailto:whl...@gmail.com]
 Sent: Friday, September 30, 2011 3:06 AM
 To: Vega Wong
 Subject: Re: [OSL | CCIE_Voice] MOH with H323 gateway
 
 
 
 Hey maybe you were 

Re: [OSL | CCIE_Voice] proctorlabs website ?

2011-09-25 Thread William Affeldt
Is anyone else having problems with proctor labs right now? 

Sent from my iPhone

On Sep 24, 2011, at 6:27 PM, edgar feliz ejfeli...@yahoo.com wrote:

 NO I trade this morning for a while and it was not available.
 
 EJF
 
 From: Rrcrumm rrcr...@yahoo.com
 To: Ken Wyan kew...@gmail.com
 Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
 Sent: Saturday, September 24, 2011 5:28 PM
 Subject: Re: [OSL | CCIE_Voice] proctorlabs website ?
 
 If it is you should send an email to supp...@proctorlabs.com  and someone 
 should be able to help you
 
 Randall
 
 Sent from my iPhone
 
 On Sep 24, 2011, at 8:34 AM, Ken Wyan kew...@gmail.com wrote:
 
 Today proctorlabs.com website is down ???
 ___
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 ___
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 ___
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Re: [OSL | CCIE_Voice] Fast Bussy

2011-08-25 Thread William Affeldt
What does your debug voice ccapi say for the inbound and outbound dial peers? 
You should be able to validate your translation and destination. Also check 
your binding.  

Sent from my iPhone

On Aug 25, 2011, at 5:57 PM, Cisco Nut rafayc...@gmail.com wrote:

 Hello
 I have setup BR2 router so that when call come from pstn to 3214-3000 it gets 
 translated to 1002 and sent to CUCM SUB / PUB, only issue is when I call I 
 get fast busy. I have attach configuration and debugs. 
 My H323 Gw is fine, I can dial from my BR1 / HQ phones to 90113214! via BR2. 
 
 Regards
 
 Trouble.rtf
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Re: [OSL | CCIE_Voice] calling through GK and CUBE - calling phone keeps ringing

2011-07-27 Thread William Affeldt
That is usually the result of fast start not being enabled. I would also try 
mtp required if that alone does not fix it. 

Sent from my iPhone

On Jul 27, 2011, at 10:52 AM, Joel Tyson joel.ty...@gmail.com wrote:

 Wait for 245 MTC is unchecked on the GK  trunk setting on CUCM.   When 
 setting CUBE to use g729 taking the transcoder out of the picture the calls 
 worked fine.  With xcoding to g711 on cube the originating phone (hq ph1) 
 continues to ring after picking up BR2 phone.Any help iseatly appreciated.
  
 Thanks!
 Joel
  
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Re: [OSL | CCIE_Voice] exam cancel

2011-07-24 Thread William Affeldt
When is your date? 

Sent from my iPhone

On Jul 24, 2011, at 9:24 AM, donny f f.faraday...@gmail.com wrote:

 hi all,
  
 does anyone has experience when need to drop off /move  lab date?
  
 I heard sick and emergency , Cisco will able to do without lose the money
  
 And what needed , how many days notice ?
  
 tks
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Re: [OSL | CCIE_Voice] unable to receive any calls on SOME IP Phones..

2011-07-22 Thread William Affeldt
There used to be a bug causing the web page to not show the phones being 
registered. I think that was in version 5 though.  I would also check srst and 
replication if you have multiple call managers. 

Sent from my iPhone

On Jul 21, 2011, at 11:07 PM, Rashid Khan me_rashid...@yahoo.com wrote:

 Hi Friends,
 
 I am facing issue with some IP Phones, these phones are able to places any 
 calls but unable to receive any incoming calls; neither internal call nor 
 calls from outside network. Also when I search for these phones in 
 CallManager 4.2 search, there status is shown is UNKNOWN rather than showing 
 IP address of Subcriber.
 
 Kindly help me, I am receiving too many calls from users.
 
 Regards
 Rashid.
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 www.PlatinumPlacement.com
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Re: [OSL | CCIE_Voice] Lab result status

2011-06-22 Thread William Affeldt
You should look at this as a positive. If they deleted your test they may just 
give you a passing grade. You will never know though. The person you should 
contact is the lab ops manager. Her name is Kathe. She is a nice person and 
will take your call at any time. 

Sent from my iPhone

On Jun 22, 2011, at 4:56 PM, George Goglidze gogli...@gmail.com wrote:

 WTF man, I can't believe this is happening to you. 
 I wouldn't be able to sleep until I got the results. 
 
 
 On Wed, Jun 22, 2011 at 10:46 PM, Pablo Meneses pmenese...@gmail.com wrote:
 Can you believe that I haven't received the answer yet!
 This has been the longest wait ever.
 I even opened a ticket with Cert Support and so far I have just received 
 templates.
 Does someone else have gone through this?
 I'm starting to think that they messed up on my exam.
 
 -Pablo Meneses.
 
 
 On Wed, Jun 22, 2011 at 2:47 AM, Julien Krieger krieger.jul...@gmail.com 
 wrote:
 Hi Pablo, 
 How did it go? Have you passed?
 
 
 2011/6/21 Pablo Meneses pmenese...@gmail.com
 Thanks Cris.
 
 Looking forward to getting my result.
 
 
 On Tue, Jun 21, 2011 at 1:23 PM, Cristobal Priego cristobalpri...@gmail.com 
 wrote:
 i know man
 i took my last attempt on a Friday as well In San Jose
 
 i didn't get my score until next Monday at 14:45 PST
 
 
 2011/6/21 Pablo Meneses pmenese...@gmail.com
 Hopefully that's true. Longest wait EVER!!!
 
 -Pablo Meneses.
 
 
 On Tue, Jun 21, 2011 at 1:19 PM, Cristobal Priego cristobalpri...@gmail.com 
 wrote:
 you will get today, wait 2.5 more hours
 that's what happens when you take it on Friday
 
 2011/6/21 Pablo Meneses pmenese...@gmail.com
 Hi experts,
 
 I am wondering how long it takes for you to received your score report this 
 days. I've taken my lab last Friday and I have NOT received the answer yet. 
 Yesterday I logged to the CCIE Online Tool and it said HOLD on the PASS/FAIL 
 field, does someone know what that means? Has someone else gone through this?
 
 I will look forward to your reply as I figure most of you understand the 
 feeling of not having the answer.
 
 Regards.
 
 -Pablo Meneses.
 
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 visit www.ipexpert.com
 
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 www.PlatinumPlacement.com
 
 
 
 
 
 ___
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 visit www.ipexpert.com
 
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 www.PlatinumPlacement.com
 
 
 
 ___
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 visit www.ipexpert.com
 
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 www.PlatinumPlacement.com
 
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Re: [OSL | CCIE_Voice] Call Preservation Issue

2011-05-31 Thread William Affeldt
Voice service VoIP
H323
Call preserve

Should be the command set. I use it as my default configuration. 

Sent from my iPhone

On May 31, 2011, at 4:04 PM, CCIE Voice ccievoicelab.c...@gmail.com wrote:

 Sorry, should have been more clear. Using H.323.
 
 The call preservation feature of H323 applies when the gateway loses 
 connectivity to Call Manager only, so that the RTP stream is preserved to the 
 phone, but here the RTP stream itself is being interrupted. 
 
 
 On May 31, 2011, at 6:58 PM, amit batra batraji...@yahoo.com wrote:
 
 What type of protocol you are using ? H323 or MGCP ?
 
 With MGCP i dont think if there is any way ..
 
 H323 can preserve call ..
 
 --- On Wed, 6/1/11, CCIE VOICE ccievoicelab.c...@gmail.com wrote:
 
 From: CCIE VOICE ccievoicelab.c...@gmail.com
 Subject: [OSL | CCIE_Voice] Call Preservation Issue
 To: ccie_voice@onlinestudylist.com
 Date: Wednesday, June 1, 2011, 3:03 AM
 
 Hello Experts
  
 An interesting problem to solve for which I'm having a hard time figuring 
 out the solution myself.
  
 We have a Call Manager node and IP Phones in one location and ingress voice 
 gateways in other location.  Both locations are connected over redundant WAN 
 links. When one WAN link fails, it takes upto 5 seconds for the redundant 
 WAN link to kick in. Issue is, if there is an active call between the 
 gateway and the IP Phone, the call simply drops when a WAN issue occurs. Is 
 there a way to preserve the RTP stream on the GW itself that can be resumed 
 once the redundant WAN link takes over?
  
 Any input is appreciated!
 
 -Inline Attachment Follows-
 
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Re: [OSL | CCIE_Voice] Voicemail not working for some calling users

2011-05-19 Thread William Affeldt
I would do a wire shark packet capture on the phone side and see if you can see 
both streams. If you can't repeat that at every hop until you find the break in 
the stream. 

Sent from my iPhone

On May 19, 2011, at 7:19 AM, Daniel Gómez danie...@gmail.com wrote:

 Hello experts,
 
 I have the following scenario:
 
 Phone A at Company A HQ(CUCM 8.0 + Unity Express 8.0) -- LAN -- ASA - 
 2Mbps WIRELESS LINK -- Checkpoint -- LAN -- Phone B at CompanyB HQ
 
 _
 
 Phone A @ Company A can call Phone B at Company B.
 Phone B @ Company B can call Phone A at Company B.
 
 PSTN users can call Phone A @ Company A.
 PSTN users can leave a voicemail to user  phone A in company A.
 
 Phone B @ Company B CAN'T leave a voicemail to user  phone A in company A.
 __
 
 At first, we were thinking about a transcoding issue but everything seems to 
 be OK. As we are limiting bandwidth to 128kbps between Company A and Company 
 B, we're using G729 codec for calls between these sites. 
 
 We have a XCODE at the gateway and it is registered correctly and seems to be 
 working OK.
 
 In the past, we have had issues regarding blocked ports on the checkpoint 
 causing 1-way audio, or Phone B not hearing PhoneA's music on hold.
 Do you think it could be part of the issue again?
 
 We're running out of ideas as we have checked everything we think could be 
 causing the issue.
 
 Thanks a lot!
 
 Daniel G.
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[OSL | CCIE_Voice] UCCX Script question.

2011-05-05 Thread William Affeldt

I am currently using a consulted transfer in a script to transfer to a a few 
phones. If they don't answer the call then it goes to Q. The issue is when the 
phone being transferred to checks it's missed call directory it shows the CTI 
port extension and not the actual calling number. Does anyone know a way to get 
it to display the actual calling number during a consulted transfer when the 
called number does not pick up. I know that I can change it to a direct 
transfer and then RNA the call back to the Q but I would like to keep the 
consult transfer. ___
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Re: [OSL | CCIE_Voice] CME Transcoder Not Working

2011-04-19 Thread William Affeldt
What does the debug h245 say? 

Sent from my iPhone

On Apr 19, 2011, at 11:27 AM, Stephen Manuel srman...@bellsouth.net wrote:

 Ladies and Gentlemen,
 
  
 
 I have the following setup in my home lab.
 
  
 
 1760 Router operating as BR1
 
 2811 Router operating as HQ w/CUE running CME 7.1
 
 2611 Router as Frame Switch
 
  
 
 I am stumped trying to get  a call from BR1 to forward to the CUE.
 
  
 
 I know CUE only speaks G711, when I force the call all the way thru at G711 
 it everything works fine, the CUE answers, I hear the recording, I leave a 
 message, MWI lights up..etc.
 
  
 
 But I am trying to have G729 used over the WAN and then transcoded into G711 
 for the CUE. I can call phone to phone using g729 with no issues, but when I 
 call phone to phone and then it rolls to the CUE, I get dead air.
 
  
 
 Any help would be greatly appreciated.
 
  
 
 Thanks
 
  
 
 Stephen Manuel
 
  
 
 Below is the pertinent config for the HQ Router.
 
  
 
 hostname HQ
 
  
 
 clock timezone EST -5
 
 clock summer-time EST recurring
 
 !
 
 dot11 syslog
 
 ip source-route
 
 !
 
 !
 
 ip cef
 
 no ip dhcp use vrf connected
 
 ip dhcp excluded-address 192.168.2.1 192.168.2.2
 
 ip dhcp excluded-address 192.168.3.1 192.168.3.2
 
 !
 
 ip dhcp pool Branch_1
 
network 192.168.2.0 255.255.255.0
 
default-router 192.168.2.1
 
dns-server 205.152.144.23 205.152.132.23
 
 !
 
 ip dhcp pool Branch_2
 
network 192.168.3.0 255.255.255.0
 
default-router 192.168.3.1
 
dns-server 205.152.144.23 205.152.132.23
 
 !
 
 ip dhcp pool HQ
 
network 10.1.200.0 255.255.255.0
 
default-router 10.1.200.1
 
dns-server 205.152.144.23 205.152.132.23
 
 !
 
 !
 
 no ip domain lookup
 
 ip name-server 205.152.144.23
 
 ip name-server 205.152.132.23
 
 ip name-server 205.152.37.23
 
 no ipv6 cef
 
 !
 
 multilink bundle-name authenticated
 
 !
 
 voice call debug short-header
 
 !
 
 voice service voip
 
  allow-connections h323 to h323
 
 allow-connections h323 to sip
 
 allow-connections sip to h323
 
 allow-connections sip to sip
 
 fax protocol cisco
 
  sip
 
   bind control source-interface Loopback1
 
   bind media source-interface Loopback1
 
 !
 
 voice class media 1
 
 media transcoder high-density
 
 !
 
 voice class codec 1
 
 codec preference 1 g711ulaw
 
 codec preference 2 g729r8
 
 !
 
 voice class codec 2
 
 codec preference 1 g729r8
 
 codec preference 2 g711ulaw
 
 !
 
 voice-card 0
 
 dspfarm
 
 dsp services dspfarm
 
 !
 
 archive
 
 log config
 
   hidekeys
 
 !
 
 translation-rule 1
 
 Rule 1 ^3 617863
 
 Rule 2 ^7 904607
 
 !
 
 interface Loopback0
 
 ip address 1.1.1.5 255.255.255.255
 
 !
 
 interface Loopback1
 
 ip address 1.1.1.1 255.255.255.252
 
 !
 
 interface FastEthernet0/0
 
 description Management Vlan
 
 ip address 192.168.1.150 255.255.255.0
 
 duplex auto
 
 speed auto
 
 !
 
 interface FastEthernet0/1
 
 no ip address
 
 duplex auto
 
 speed auto
 
 !
 
 interface FastEthernet0/1.10
 
 description HQ Servers
 
 encapsulation dot1Q 10
 
 ip address 10.1.200.1 255.255.255.0
 
 !
 
 interface FastEthernet0/1.700
 
 description HQ Phones
 
 encapsulation dot1Q 700
 
 ip address 10.2.200.1 255.255.255.0
 
 ip helper-address 10.1.200.21
 
 !
 
 interface Serial0/3/0
 
 description Frame Relay to BR1 and BR2
 
 no ip address
 
 encapsulation frame-relay
 
 frame-relay lmi-type cisco
 
 !
 
 interface Serial0/3/0.101 point-to-point
 
 description BR1
 
  ip address 172.16.1.1 255.255.255.252
 
 frame-relay interface-dlci 101  
 
 !
 
 interface Serial0/3/0.201 point-to-point
 
 description BR2
 
 ip address 172.16.1.5 255.255.255.252
 
 frame-relay interface-dlci 201  
 
 !
 
 interface Service-Engine1/0
 
 ip unnumbered Loopback1
 
 service-module ip address 1.1.1.2 255.255.255.252
 
 service-module ip default-gateway 1.1.1.1
 
 !
 
 router eigrp 100
 
 network 1.0.0.0
 
 network 10.0.0.0
 
 network 172.16.0.0
 
 network 192.168.1.0
 
 no auto-summary
 
 !
 
 ip forward-protocol nd
 
 ip route 0.0.0.0 0.0.0.0 192.168.1.1
 
 ip route 1.1.1.2 255.255.255.255 Service-Engine1/0
 
 ip http server
 
 no ip http secure-server
 
 ip http path flash:gui
 
 !
 
 nls resp-timeout 1
 
 cpd cr-id 1
 
 !
 
 control-plane
 
 !
 
 voice-port 0/0/0
 
 description HQ Router Analog Phone
 
 station-id number 9046074899
 
 !
 
 voice-port 0/0/1
 
 !
 
 voice-port 0/0/2
 
 !
 
 voice-port 0/0/3
 
 !
 
 sccp local Loopback1
 
 sccp ccm 1.1.1.1 identifier 1 version 7.0
 
 sccp
 
 !
 
 sccp ccm group 123
 
 bind interface Loopback1
 
 associate ccm 1 priority 1
 
 associate profile 1 register mtp001b542cb0f1
 
 associate profile 2 register CFBridge
 
 keepalive retries 5
 
 switchover method immediate
 
 switchback method immediate
 
 switchback interval 15
 
 !
 
 dspfarm profile 1 transcode 
 
  codec g711ulaw
 
 codec g711alaw
 
 codec g729ar8
 
 codec g729abr8
 
 codec g729br8
 
 codec g729r8
 
 maximum sessions 4
 
 associate application SCCP
 
 !
 
 dspfarm profile 2 conference