I am a CCIE Collaboration # 26244 and over the years I found that it is not easy to find a solution when you are stuck in a real world problem. Theoretical stuff is very different when you face something in a live environment. Entirely a different ball game! You may have to go through a ton of documentation, admin guides, logs etc. in order to understand the issue and then work on the problem. Sometimes you don’t even have all that time to spend reading and understanding the issue due to the critical nature of modern-day businesses. Customers expect to get their problems resolved in a blink of an eye.
This is where the idea of writing my very own experience and sharing knowledge came to my mind. This platform will help you get to the bottom of an issue real quick and I hope it will save you a lot of time. You will find how I came across a problem and how I managed to resolve it keeping in mind the technical perspective. I have also added some useful content where I have tried to make complex collaboration stuff simple!
If you have come to this platform searching for a resolution of your problem or to read an article and if it has helped you in any way then I would be very happy if you leave a comment for me and subscribe. This will at least keep me motivated to keep this blog running!
Any suggestion or any question on my entries in this blog are most welcome and I always try my best to answer all your queries.
Thanks,
Ash.
voiceonbits.com 
Hi Ashar
Nice work mate !!!!
i was attempt for the voice lab 3 times and every time was different …
the result sill fail !!!!
and i did the last one on 23th of June , which is you just post the “MGCP Trouble ” on 22nd of June …
this question i did not complete it as i was use debug mgcp input .. and after that near kill my HQ router ..
but good to see your solution in here ..
so did you passed the voice lab for this lab ?
Hi Justin,
Thanks for your feedback.
I have been sharing solutions of what I use to come across in my day to day work. Most of it has been requested by my friends as you will not be able to find a comprehensive solution at one place for most of the things. I had to study different Cisco documents to put everything together.
Quick hint for the MGCP debug – before you start ‘debug mgcp packet’ you must login to Putty session so that it saves the session. Please no questions regarding lab for reasons you very well know 🙂
I will soon be writing CCIE Voice lab tips which may be helpful for all Lab takers.
Thanks again and do let me know if you have any suggestions.
Hey Ashar, nice seeing your blogs, it’s really great though just found the URL now, let me bookmark it. 🙂
“I will soon be writing CCIE Voice lab tips which may be helpful for all Lab takers.
Thanks again and do let me know if you have any suggestions.”
Yes, I’m waiting for the lab tips and I wish to know about the strategy you followed. 🙂
Thanks! Mijanur Rahman
dreamforccie.wordpress.com
This is an excellent post and may be one that ought to be followed up to see what goes on
A chum sent this link the other day and I’m excitedly hoping for your next content. Carry on on the first class work.
thank you for nice and interesting blog! now i know where i can find information!) cheers from russia)!
I have one question in my mind running since long time. If any on e can help me in this….
On HQ site globalization and localization is done…. So by hitting directories missed/received calls and dial, calls go through……
But in Site B and Site C, when hitting directories missed/received calls and dial………….(for emgr, local and intrl calls) The call should succeed or not.
Because for calls to succeed prefix 9 should be done.
Does the mark get deducted in the lab exam if the calls are not working in this way….
@Royal – You will need to configure as per the requirement. If the requirement is that HQ users should be able to dial back from directories then you will configure it for HQ only, no need to do it for Site B and Site C if it has not been asked. If they ask that Site B and Site C users should also be able to dial back from directories then you will have to do it for both these sites as well. Please keep yourself confined to what has been asked, don’t get into configuring nice-to-have stuff as it will not fetch any points but it may take up your precious time which you will come to know by the end of the lab. You will not lose any points for not configuring anything which has not been asked. This is what I understand, offcourse I am not Cisco and cannot speak on behalf of them.
Hello
i am trying to implement cme with cue but i faced a problem . the problem is when i change the internal extensions to 3 digits format auto attendant do not forward calls to internal extensions Although i changed all extension in cme and cue to 3 digits format and changed the dial plan in cme which points to cue . what else i should do to get cue aa forward calls to internal extension
thanks
Shehab, It should ideally work if you add proper extension – make sure you add dial peer which send incoming call to CUE
i dont knew i think auto attendant script is programmed to deal with 4 digits length extensions
Ash,
excellent site, how about an ACCURATE how to for setting up extension mobility ?
Hi Strick, I have foused on complex issues on this blog but will definitely work on “How to” for Extension mob.
Thanks for feedback.
Hi, thanks for very useful information from VOIP world, I hope you will be continue with sharing interesting scenarios, thanks again.
Hi,
Could you please show some lights on ICT between callmanagers in different locations
Hi ash,
Thanks for sharing, really nice site !
Help me a lot.
Hi Ashar,
I would like to thank you for the useful information, God with you Dear.
Hi guys , i wanna ask something , can u please tell me in asia how we set up isdn Pri , if we are using it on mgcp gateway or other gateway?
Please look at this link:
https://voiceonbits.com/2010/06/21/mgcp-gateway-setup/
Highly appreciate what you offer over here.
Thanks !!
HI ALL
i have one question please tell….
there are 100 phones in a cluster
all phones are registered to same server.working fine……but sudden 50 phones
goes on registration process and go on ……those 50 phones are not able to register
what might be issue…?
Thanks
Dharambir
HI Dharambir, Are these phones in same location? check the network settings for the ones which are not working and try to do a factory reset? Are you using DNS? This suggest a network issue to me so you should check the switch where these phones are connected..
Hi
I need to configure a number with auto-answer and a music should be playing all the way from beginning to end. when we dial the number it will be auto-answer and music from MOH should be playing background continually. I heard this can be done with scripting. but have no idea. even some documentation to follow will be helpful.
I have CUCM 8.5 and Cisco Unified CCX
I am new to voice. I need a proper guidance for this..
Hi all,
Im preparing my ccnp voice and im really getting confused when configuring dial-peers. can anyone help me clarify the below: i believe the below is an incoming dial-peer but how come there is an outgoing translation pattern as well?
dial-peer voice 1 pots
preference 1
destination-pattern 0T
translate-outgoing calling 2
incoming called-number .
direct-inward-dial
!
one more thing, what does that match destination-pattern 12345T
Hi Joe
I understand the confusion as dial-peers are a bit complex to grasp. I have written about dial-peers as well on this blog and would be helpful. Please see this:
https://voiceonbits.com/2010/07/04/h323-dial-peers-made-simple/
Regarding your question, this dial-peer is acting as an outbound as well as inbound POTS dial-peer. The direct-inward-dial & incoming called number . command means incoming call on E1 will come through this dial-peer and then will match an outbound VOIP dial-peer to go to Call manager, For outgoing calls from CCM to PSTN, calls will go out through this outbound POTS dial-peer after match that 0T pattern. The translate-outgoing will change the calling number so if you have 4 digit extension, it may change it to full PSTN number (depends how your provider would like to see).
The destination-pattern 12345T means any dialed number which starts with ‘12345’ and then any number of digits will match this pattern and all digits will go to PSTN except ‘12345’ as they will be match explicitly. You may need forward-digits all or prefix 12345 to send 12345 as well.
Hope this helps.
Thank you asharsidd, that was very helpful. I have tons of questions 🙂
please tell me why we call SIP a SIP trunk. why no H.323 trunk and MGCP trunk???
Hi asharsidd,
Good morning, i would like ask your help for my problem. I can not connect my web page Cisco Unified Communications Manager Administration.
When i using my ip address its open the main web page, after that i try to click the link Cisco Unified Communications Manager Administration, to use my username and password and cant not open.
Please, give me any tip
Best Regards
Al Monte
Hello, If I understood correctly you are getting an error when you are entering you ccm admin username password. If you are getting a database communication error then I would suggest restarting TFTP and if necessary restart CUCM but if you are not getting any error then you may be entering wrong username and password. You may reset admin username and password by logging in through CLI. You may find that method on this blog if you need.
Hi asharsidd,
No, the problem is that i can not open the link page Cisco Unified Communications Manager Administration, because this is the page where i can use my username and password to login in cisco callmanager admin page.
Best Regards
Al Monte
Your statement is bit different of what you said initially.
If that is the case then restart tomcat service from CLI
utils service restart Cisco Tomcat
Hi asharsidd ,
I have tryed this command before, and the service Tomcat only stopped, and didnt restart. After that, i have applied the command utils service start Cisco Tomcat, just to start the tomCat service again.
Best Regards
Al Mont
Hi asharsidd
admin:utils service restart Cisco Tomcat
Service Manager is running
Cisco Tomcat[STOPPING]
Cisco Tomcat[STOPPING]
Cisco Tomcat[STOPPING]
Cisco Tomcat[STOPPING]
Cisco Tomcat[STOPPING]
Cisco Tomcat[STOPPING]
Commanded Out of Service
Cisco Tomcat[NOTRUNNIG]
Service Manager is running
Cisco Tomcat[STARTED]
I have done now this command like you can show above, and untill now, i can not open the page.
Best Regards
Al Monte
Hi asharsidd
there is another service that stopped:
Connection Database Proxy[STOPPED] Commanded Out of Service
What you can say about this service, because another one are running
Best Regards
Al Monte
Connection Database Proxy[STOPPED] Commanded Out of Service
That service cannot be at CUCM..are you looking at CUCM or Unity connection?
See if you can ping the IP address of CUCM. Can you get to GUI of a subscriber? I would go for a clean cluster reboot if problem persists.
Yes, i can ping to the router and server also, the ping works, but i can not get in to the GUI.
I cannot access into Cisco Unified Communications Manager Administration, because the page cannot be displayed.
Best Regards
Al Monte
Hi asharsidd,
When i try to use my Real-Time Monitoring Tool, to check what is happend, im receiving the following message:
RTMT application cannot communicate with specified callmanager cluster(200), please verify the host ip address is correct and network connection is UP, and try again.
Best Regards
Al Monte
Try from internet explorer, firefox. Also add CUCM page under internet options >security > sites
You sure you are using correct format like https://IP/ccmadmin ??
asharsidd,
yes, im sure, and im one of the administrators, because the rest also found the same problem.
Best Regards
Al Monte
Hi asharsidd,
Yesterday, i also found this bug, and now i think that this can be one of the issues to get my web page admin down.
HTTP Status 404 – /ccmuser/showhome
——————————————————————————–
type Status report
message /ccmuser/showhome
description The requested resource (/ccmuser/showhome) is not available.
——————————————————————————–
Apache Tomcat/5.5.17
Best Regards
Al Monte
Why people still use to read news papers when in this
technological globe everything is presented on net?
Thank you for some other excellent post. Where else may just anyone get that type of info in such an ideal means of writing?
I have a presentation subsequent week, and I’m at the look for such information.
hi sir , i have 4 fxo line in case of h323 gateway
and i have 4 internal extension
1001
1002
1003
1004
it is possible 1001 can use line 1 for incoming and outgoing
1002 use line 2 for incoming as well as outgoing .
same thing for other 2 extension
Hi Sushant, yes you can. In CCM when you add FXO port, under Port information you can select “bothways” to use FXO port for both inbound and outbound.
H323 gateway you can.t see bothway
What is your gateway and what module you are using in its slot?
I am using h323 gateway . module name vic2-4fxo card . I have 4 fxo line , internally I have 4 ip phones. I am trying to implement phone 1 can use inbound and outbound calling from line 1 .and same other extension 2 use line 2 respectively . in this gateway we have already 8 fxo line , 1 pri link total 12 fxo line and 1 pri ….but my concern only 4 fxo line
I will ask you again..which gateway is this? 28xx? 29xx? 39xx?
2911
How are you getting the option for FXO on a SCCP 2911? what module you are selecting? I can only get FXO when I make the gateway as MGCP.
Hi dear i am facing very serious issue , in case of srst my sip phone not registering , sccp phones working fine. please help
You may have missed the SIP commands on the gateway, please add following and reload gateway and it should work.
voice register global
max-dn 10
we need to type only 2 commands it will be work , can u please give me full configuration,
voice register global
max-dn 10
thanks in advance
Hi asharsidd, I just want to thank you! This is an awesome blog! Nice work and I hope I bring you enough traffic to keep it going for you.
dEAR I am facing one issue Tomcat service is not starting in cucm. Pleae helod
Hi,
my name is yogesh Sawant, i am working on Panasonic PBX, one of my customer need to use AVAYA 1608I and Cisco phones CP 7821 on Panasonic pbx, can anyone give me suggestions how to use this phones on Panasonic pbx
Thanks
Yogesh Sawant
+91 9004242551
HI ASH,
I JUST STARTED READING YOUR BLOG ” H323 Dial Peer made simple” AND HAVE DOUBTS REGARDING TWO POINTS:
1)FOR “Outbound Call from CUCM to PSTN:” —————->This is what a gateway does to match an inbound VoIP dialpeer—————>IN 3RD POINT YOU’VE MANTIONED ” Match the Calling number (ANI) with destination-pattern”…
CORECT ME IF I’M WRONG, THIS SHOULDN’T HAVE BEEN ” MATCH THE CALLED NUMBER(DNIS) WITH DEST-PATT” ???COZ, Dest-patt ALWAYS POINTS TO DIALED NO AND NOT CALLING NO. ???
2)ALL THE ABOVE SAME FOR “Inbound call from PSTN to CUCM” ——–>This is what a gateway does to match an inbound POTS dialpeer:>>>>>>>>>>IN THIS SAME 3RD POINT:
Match the Calling number (ANI) with destination-pattern ?? DROM MY VIEW THIS ALSO SHOULD BE SAME THAT’S “CALLED NO WITH D-P”…
PLEASE END MY DOUBT ..COZ I’VE FOUND YOUR BLOG AN INTERSTING ONE AND WILL DEFINITELY BE MOVING ON TO OTHER TOPICS OF YOUR BLOG.
THANKS.
Dear Amrik, Sorry for late response. For both of your points I would say that when an inbound call legs are matched then destination-pattern match ANI (Calling number) and when an outbound leg is matched then same destination-pattern looks for DNIS or called number. Hope this helps.
Hi
I have on problem i registered one ip phone in CME, Ring going but when he picked up his ip phone the beef sound will come, call will disconnected!
any info
Hi Rasool, it would be difficult to tell from your problem description as I am not sure you talking about SIP or SCCP? Are you using ISDN/SIP/FXO? Are the phones local to the CME? What codec is in use? Is there a transcoder setup? Are you using G711u or G711a? Make sure you configure a transcoding resource on router and point CME to it.
dspfarm profile 2 transcode
codec g711ulaw
codec g729abr8
codec g729ar8
maximum sessions 2
associate application SCCP
codec g711ulaw
session protocol sipv2
@ Rasool – You still have not answered all my questions. Are you using voice register global or telephony service? is the phone SCCP or SIP etc? I would recommend you go through one of my entry on basic CME setup. You must be missing something in the configuration and it will help.
https://voiceonbits.com/2010/07/06/cme-basic-setup/
Also during an active call run a debug voice ccapi inout, debug ephone all and see what is happening.
Hope all you doing good,
In My office Voice network there is one is going on, Can some one help me to fix the issue please.
If call to 10 digit number it asking me to hit 0, Once i hit 0 after one ring call is disconnecting, But i call 4 digit number it is working fine, Can some one help me to fix this issue.
What is the difference bettween CUBE and SBC?
Can You please help me with the detailed view on SIP Trunk