gomez’s blog

In following blog posts I will write about different Cisco based VoIP solutions and I will try to cover some foundation topics like voice signaling protocols and so on. My aim is to present different solutions and possibilities and I hope that you will find it interesting and helpful.

About MGCP and reasons to use it

Media Gateway Control Protocol is client server voice signaling protocol. Call control is handled by call agent (Cisco Unified Communications Manager) and media translation is performed by voice gateway. It is important to mention that it is only client – server model based voice signaling protocol on the market. Also, it is well know for its simple configuration (it is just matter of configuring call agent and remembering several gateway commands), centralized management and one of the main benefits is it’s use for MGCP back-haul QSIG configurations (in short: Layer 2 signalization is handled by voice gateway which is under control of call agent, and layer 3 signalization is forwarded to call agent – Cisco Unified Communications Manager – CallManager). Please note that MGCP is often called Megaco by telco people.

MGCP Messages

1. Create connection – CRCX
2. Notification request – RQNT
3. Notify – NTFY
4. Modify connection – MDCX
5. Delete connection – DLCX
6. Audit endpoint – AUEP
7. Endpoint configuration – EPCF
8. Restart in progress – RSIP

MGCP Call flow

MGCP configuration basics

This part of text will cover MGCP configuration – just foundations. Please note that it won’t include all specific configurations, it will just cover base configuration, but in several new posts I will try to cover different configuration scenarios and solutions.

So, let assume that MGCP call agent, in our case Cisco Unified Communications Manager – CallManager is working on IP address 192.168.1.95. Also, assume that redundant CallManager is working on IP address 192.168.1.100.  This text does not show CallManager side of configuration of a gateway and I assume that that part is already successfully configured (I will eventually edit this blog post to include that configuration, afterwards)

Configurations is as follows:

1. From global configuration mode start MGCP proces:

Router#conf t
Enter configuration commands, one per line.  End with CNTL/Z.
Router(config)#mgcp
Router(config)#

2. Next thing we need to do is to configure call agent, redundant call agent and we need to inform voice gateway that call agent is under control of MGCP. We can do that as follows:

Router(config)#mgcp call-agent 192.168.1.95
Router(config)#ccm-manager redundant-host 192.168.1.100
Router(config)#ccm-manager mgcp

3. Once done with that, we can configure out Cisco Unified Communications Manager gateway options. So to do that, navigate to your CallManager and in Cisco Unified CM Administration configuration menu select Device > Gateway > Add new. Select your voice gateway model from the drop down list and click Next. For Protocol select MGCP and click Next. Once there, configure Domain name (please note that this field needs to be in following format: gatewayhostname.domainname, for example CCIEVoiceLab.localnetwork.lab) and other required options (add cards and preform complete gateway configuration). Once you are done with voice gateway configuration you are required to configure dial plan which will include this gateway into configuration. That goes beyond the scope of this blog post and I will write about it in some of following blog posts.

4. Once done with CallManager configuration we need to instruct voice gateway about the TFTP address where CallManager stores configuration, and that is typically IP address of the CallManager it self:

Router(config)#ccm-manager config server 192.168.1.95
Router(config)#

5. Once we are done with that we can instruct our gateway to download and apply configuration by issuing following command:

Router(config)#ccm-manager config
Router(config)#

Wait several seconds and you can verify your downloaded and applied configuration with following command:

Router#show running-config
Router(config)#

At that point your voice gateway will be controlled by call agent, in this case CallManager. To verify your configuration please give a check to Troubleshooting commands.

MGCP Troubleshooting

Once you are done with MGCP gateway configuration, you can verify your configuration by issuing following commands:

Router#show mgcp
MGCP Admin State ACTIVE, Oper State ACTIVE – Cause Code NONE
MGCP call-agent: 192.168.1.95 Initial protocol service is MGCP 0.1
MGCP validate call-agent source-ipaddr DISABLED
MGCP validate domain name DISABLED
MGCP block-newcalls DISABLED
MGCP send SGCP RSIP: forced/restart/graceful/disconnected DISABLED
…

This command will show all MGCP settings, timers and so on.

Router#show ccm-manager
MGCP Domain Name: Router
Priority        Status                   Host
======================================
Primary         Registering with CM      192.168.1.95
First Backup    Down                     192.168.1.100
Second Backup   None

Current active Call Manager:    None
Backhaul/Redundant link port:   2428
…

This command will show registration status and other statistics.

Beside above commands, we can use:

Router#show mgcp endpoint
Router#show voice port summary

That would be all for this post. I hope that this blog post is helpful and if you do have some questions in regards to MGCP please let me know and we can try to find solution. In next several posts I will write about MGCP fallback, backhaul and DTMF configurations.

1. Media Gateway Control Protocol
2. MGCP messages

Two years ago I have decided to start working on Cisco Voice Professional certification track. It was logical step to make for me since I was working for several years in voice industry with different solutions, so when I started with Cisco solutions there was no doubt what to choose and why. In prior to further discussion about certification I would like to point that I am full time employed with lot of different responsibilities which can and does impact preparations for certification.  So, for some people that might take this certification track this might be resolved faster, but all depends on attitude, basic knowledge and many other factors. I took a newer track based and created around Cisco Unified Communications Manager 6.0. When I started preparing myself for CCVP, there was active certification based on Cisco Unified Communications Manager 4.1, and main difference beside versions of Cisco Unified Communications Manager  is that old certification path included Gateway and Gatekeeper exam. I took all of the exams as Cisco is recommending on their web site for CCVP certification. So, that is as follows:

Cisco Certified Network Associate Voice – CCNA V

I was surprised when I tried to place a VoIP call over UMTS/3G network using my Nokia E72. It wasn’t working. In prior to that, I needed to install Nokia SIP VoIP settings application to get VoIP support on my mobile phone (Nokia E71 comes with this software preinstalled, so if you are Nokia E71 user and possible switcher to Nokia E72 be aware of this). I was really positively surprised when I saw how many different options can you configure using Nokia SIP VoIP settings application (for example, you can change your QoS DSCP value for RTP traffic, RTP port range, codecs and so on… impressive.) However, my VoIP calls placed over 3G were unsuccessful. So, I needed to find solution for it.

As a prerequisite make sure that you are connected to UMTS network by navigating to:

Home > Control Panel > Settings > Phone > Network > Network Mode

While there for “Network Mode” field you will need to set “UMTS” and once you apply that you will be connected to your UMTS/3G network (in upper left corner you will be able to see small 3G connection mark).

Once done with that we need to enable 3G VoIP calls.

Procedure is as follows: Navigate to Home > Control Panel > Net Sett:

Once there, navigate to “Advanced VoIP settings” and then select “VoIP services“:

After that you will need to select your SIP profile (which is in our case called devlogic) and after that select “Profile settings” as per screenshot:

Once there, you will need to scroll down and find option “AWCDMA“. That option is set to “off” by default, and you will need to set it to “on” in order to place VoIP calls over 3G network.
W-CDMA (Wideband Code Division Multiple Access), UMTS-FDD, UTRA-FDD, or IMT-2000 CDMA Direct Spread is an air interface standard found in 3G mobile telecommunications networks. Once done, you are ready to make your UMTS/3G VoIP calls.

SIP VoIP settings
Installing SIP VoIP settings
AWCDMA

Few days ago we finally got our new optical connection. It is working like a charm. Having optical connection and Cisco router from one side and Asterisk server from the other side was a clear sign that we could/should implement QoS for voice traffic. By default, Asterisk is using port range 10000 to 20000 for RTP streams (which is adjustable in /etc/asterisk/rtp.conf) so you have several solutions how to implement Low Latency Queuing – LLQ on Cisco router. You can:

1. use Network Based Application Recognition NBAR to recognize your RTP traffic by using command ‘match protocol rtp’ within required class map;
2. create access control list which would comprehend all the traffic which is working as UDP in port range 10000 to 20000 by executing command ‘access-list 101 permit udp any range 10000 20000 any’ and then call that access control list within class map by using command ‘match access-group 101′;
3. make your Asterisk mark your SIP or RTP traffic by default.

From above listed solutions to this problem, third solution was most logical to me, although all solutions would work. Since there is recommendation to mark traffic as close to source as possible third option was very logical thing to go for. If you are using Asterisk which is working on Linux as root user (in most cases it is working with asterisk user by default) you will need to edit your sip.conf and within section [general] you will need to add following:

tos_sip=cs3
tos_audio=ef

Once done with that, save your configuration, reload asterisk and you are ready to go for Cisco configuration.

As I already mentioned, in most cases Asterisk is using user asterisk for the Asterisk daemon. If that is a case, above listed solution won’t work for you because Linux as operating system won’t allow you to mark your packages as non root user. That is very logical since we would be able to mark our web or some peer to peer traffic packages as EF (expedited forwarding) and obtain priority which is not desired at all. But, there is very powerful solution by using iptables. As root, you will need to execute following:

iptables -A OUTPUT -t mangle -p udp -m udp –dport 5060 -j DSCP –set-dscp 0×28
iptables -A OUTPUT -t mangle -p udp -m udp –sport 10000:20000 -j DSCP –set-dscp 0×28

By executing above commands you will mark your SIP and RTP traffic as DSCP CS5 (IP Precedence 5). You can verify that by catching a trace using snoop/tcpdump.

In case that you would like to set different DSCP values for some traffic, please find partial list here:
Expedited Forwarding, DSCP = 0x2E
CS5, DSCP = 0×28
CS6, DSCP = 0×30
CS7, DSCP = 0×38

Now last thing that we would need to do on the Linux side is to add those iptables rules to load on boot. Edit /etc/rc.local using your favorite text editor and paste above listed iptables lines and save it. Please note that rc.local file is loaded after the network, so if you are planning to put some firewall lines beside those that mark sip and trp traffic you might end up without your firewall rules for few seconds (otherwise, you can execute ‘iptables-save >/etc/iptables.rules‘ and then you would just need to add following line ‘pre-up iptables-restore < /etc/iptables.rules‘ to ‘/etc/network/interfaces‘ by using your favorite text editor to be loaded with network).

On the Cisco router side, we will configure Low Latency Queueing – LLQ to put some priority onto our CS5 marked packages. First thing that we need to do is to create class map from the global configuration mode by executing following commands:

router>enable
router#conf t
router(config)#class-map match-any voice
router(config-cmap)#match ip dscp cs5
router(config-cmap)#exit
router(config)#

By creating class map as listed above we would select our Asterisk marked cs5 packages. Next thing that we need to do is to define what we want to do with above selected cs5 marked packages. We are doing that by creating policy map as follows:

router(config)#policy-map llq
router(config-pmap)#class voice
router(config-pmap-c)#priority percent 20
router(config-pmap-c)#exit
router(config-pmap)#class class-default
router(config-pmap-c)#fair-queue

As per above configuration we are creating policy map called llq, and within that we are specifying 20 percent of available bandwidth for class voice. Please note that we need to have correct bandwidth statement under interface which is connected to WAN. Also, please note that we have enabled fair queueing on class-default (all data which is not selected by some other class map).

Last thing to do in regards to configuration is to enable policy map on interface which is facing connection towards your Internet Service Provider (WAN interface). We can do that by navigating to interface configuration mode and applying service policy in output direction as per following example:

router(config-pmap-c)#exit
router(config-pmap)#exit
router(config)#interface fastEthernet0/1
router(config-if)#service-policy output llq

And that is all. Now, we need to check results of our work by executing following command:

router#show policy-map interface fastEthernet 0/1
FastEthernet0/1

Service-policy output: llq

queue stats for all priority classes:
Queueing
queue limit 64 packets
(queue depth/total drops/no-buffer drops) 0/0/0
(pkts output/bytes output) 947031/193919404

Class-map: voice (match-any)
856803 packets, 170900643 bytes
1 minute offered rate 22000 bps, drop rate 0 bps
Match: ip dscp cs5 (40)
856803 packets, 170900643 bytes
1 minute rate 0 bps
Priority: 20% (400 kbps), burst bytes 15000, b/w exceed drops: 0

Class-map: class-default (match-any)
5321936 packets, 1534281066 bytes
1 minute offered rate 20000 bps, drop rate 0 bps
Match: any
Queueing
queue limit 64 packets
(queue depth/total drops/no-buffer drops/flowdrops) 0/0/0/0
(pkts output/bytes output) 5322316/1531545858
Fair-queue: per-flow queue limit 16

Please note packet counts under voice class on this policy. That means that we have deployed working configuration which will improve your voip experience. Your voip traffic will get priority among other aggressive traffic flows. Please find some additional information below, and feel free to contact me in order that you have additional questions.

1. Low Latency Queueing
2. Quality of Service

More than year ago when I started using Nokia Eseries one of the reasons to switch to new mobile platform was SIP stack and client support with Eseries (I used to have Nokia E71, now I am proud owner of Nokia E72). By that, I was able to connect to Asterisk or Cisco Unified Communications Manager (by using SIP digest authentication) and that was working like a charm (I am still using SIP integration heavily).

Then, I started to think about different approaches with Nokia in business environment, followed with different cost saving strategies so i asked myself about Nokia Eseries integration with Unified Communication Manager (UCCM and CME environment) by using SCCP. That was logical thing to think of, since there is SIP support already integrated into this phone series and there are plenty of Call Manager deployments in production worldwide natively working with Skinny (SCCP stands for Skinny Client Control Protocol which is often just called Skinny). And, as expected, there was really nice integration prepared by Nokia for their business users called Nokia Call Connect for Cisco.

Nokia Call Connect for Cisco integrates compatible Nokia Eseries devices with compatible enterprise voice infrastructure. When you enter the coverage area of the office wireless local area network, your device automatically registers to Cisco Unified Communications Manager and thereby activates business mode. In business mode, you can use Cisco Unified Communications Manager services to handle business calls.

With Call Connect, you can:

• Use high-speed WLANs instead of cellular networks to make calls when you are within WLAN coverage
• Use the services of Cisco Unified Communications Manager to handle business calls
• Route calls over the enterprise voice and data network to help minimize mobile phone bills
• Benefit from improved mobile phone coverage within buildings by using high-speed WLANs
• Receive notifications of new voice mail as text messages
• Access online services, such as corporate directory

Solution deployment can be splited in two parts:

• Configuring Unified Communications Manager
• Configuring Nokia Eseries phone

Since I am working in lab environment with Unified Communications Manager 6.0 and Nokia E72, I needed to make sure to download proper required packages, as follows:

Nokia Call Connect for Cisco client v2.0 (v2.0(1005)) (SIS, 1,66 MB)
NOTE: Supported devices include Nokia E52, Nokia E55, Nokia E63, Nokia E66, Nokia E71, Nokia E72 and Nokia E75

Cisco option package (COP) file for CUCM 4.x, 5.x and 6.0 for Nokia Call Connect for Cisco clients (.zip, 8 kB)
The file should be imported to Cisco Unified Communications Manager server 4.x, 5.x and 6.0 to add the Nokia Eseries devices in the device list of Communications Manager if the correct device type isn’t yet included there.

Please note that client file provided in the list above is actually trail version of the Nokia Call Connect for Cisco, and it is going to be active for 60 days after which it is going to expire and you will need to purchase real license. This text is not going to describe how to install this client, but I will suggest to install it from OVI store (it is easiest and most convenient way to do so).

Cisco Unified Communications Manager Configuration

After we have downloaded above files, we need to import Cisco Option Package to Unified Communications Manager. Once we are done with that process we will have Nokia S60 listed as valid phone type in CallManager Phones configuration. Process of adding COP file is straight forward and is described in following sections. Please note that you will need to have up and running FTP server in your network to accomplish installation of COP file.

First, we need to navigate to Cisco Unified Communications Manager Serviceability configuration pages (selection can be made from upper right corner as shown on screenshot):

Once logged into Serviceability configuration pages, we need to navigate to Software Upgrades drop down menu, and we need to select Install/Upgrade:

Once there, we need to assume control if there was some previous session:

Next thing is to select preferred source of installation. In our case that is going to be FTP server. Valid options are Remote Filesystem and CD/DVD.

Please fill all required fields (fields indicated with *). Also, make sure to put COP file in root directory of your FTP server, so that CallManager is able to find it as valid upgrade option. You will need to provide Directory (put / for root on your FTP server), Server (IP address of your FTP server), Username and Password (valid user information) and Transfer protocol which can be SFTP and FTP.

Once done with filling up required field, press Next and Call Manager will attempt to contact FTP server. If there is valid COP file (valid upgrade option), and if we are working with proper user information and running FTP server, Call Manager will list valid upgrade options  as per following screenshot:

Please note that valid COP file for Nokia S60 phone type is called cmterm-nokia_s60_001-sccp.cop.sgn. If that is what you have listed, press Next and importing process will start (downloading):

Once it is downloaded you will be presented with MD5 hash value which you can compare with one provided by Nokia from security reasons:

Once you press Next, import process will start and you will be presented with progress as follows:

Process will run for few minutes and once it has been completed, you will be prompted about it as follows:

Once done with this step, we have imported new phone type to Call Manager: Nokia S60. To make sure that it is there, we will need to check it in Cisco Unified CM Administration configuration pages. Therefore, we will need to make proper selection in upper right corner:

Once we are in Cisco Unified CM Administration pages, we need to navigate to Device drop down menu and we need to select Phone.

Once there, we will need to add new phone, and we can do that by pressing Add new button as per following picture:

Once the page is loaded we need to make proper selection, and in our case we need to select Nokia S60. If we are able to see Nokia S60 then our COP import was successful.

Now, assuming that you have installed Call Connect client (.sis) on your Nokia Eseries device and that it works fine, we can continue with configuration. In this section, we will add new phone with phone type device Nokia S60 as listed above. Please notice that product type is now listed as Nokia S60 and that Device protocol is Skinny:

Next thing that we need to do is to check  wireless MAC address on phone since it is one of the required fields in order to add new phone. Fastest way to achieve that on Nokia E71/E72 is to type following code: *#62209526# and you will be prompted with WLAN MAC address. Once you type that into MAC address field, Description field will be populated automatically. Please note that we need to fill up all the fields indicated with asterisk (*).

We will needed to select Phone Button Template and Commong Phone Profile fields since they are required. Also, that includes Presence Groups and Device Security Profile fields as shown on following screenshot:

Once done with basic configuration, we need to save changes by hiting Save button. Then, we need to configure associated information which includes configuration of directory numbers as per following:

Once there, we need to click on Line [1] – Add a new DN and we need to fill up required fields. Field of interest is Directory number. Also, please notice that in Associated Devices box, our Nokia Eseries device will be listed:

Once done with directory number configuration, click on Save and you will have your new Nokia S60 phone configured and listed on phone list, as follows:

Once we are done with configuring Cisco Unified Communications Manager side which includes importing COP file and configuring new phone, we can start configuring Call Connect client which we installed on Nokia Eseries phone.

Configuring Nokia Eseries phone

Since we have server side up and running, we can start configuring our Nokia Eseries Call Connect client. In our lab environment, we are using Nokia E72 and following screenshots are taken on that phone. First thing that we need to do is to open installed application by navigating to Menu > Apps > Nokia CC Cisco. Please notice that Call Connect is offering multiple productivity features such as Call pick up, Group Call pickup, Call divert and DND. Also, please notice that we do not have SCCP active profiles, and in following sections we will describe process of configuring one.

To configure new SCCP service, select Options > Settings > New profile

Once in the New profile configuration mode, we will need to configure Profile name, select default Access Point and configure TFTP server. Please note that in our example Profile name is set to Call Manager, 6BFlat5 is default access point and that TFTP is manually set to 192.168.1.10 which is in our case IP address of our lab Call Manager. Valid option for selecting TFTP server is also DHCP, but in that case we would need to configure DHCP server with option 150, which would indicate IP address of our TFTP server.

Once we are done with basic profile configuration, we can click on Back. We will see our new profile in “Not registered” state. To register service, we will need to navigate to Contacts and then from viable options drop down list we need to select Cisco VoIP and select “Activate service”, as shown below:

Once you activate the service, you will see your newly created profile registered.

Once registered, one additional step can be made in order to make sure that all is working properly. Navigate to Menu > Apps > Nokia CC Cisco and select Status information. You will be able to check what is the Stack version, Outgoing phone number, License information, MAC address, DHCP related information, Networking information and SCCP profile information.

To make sure that all is running fine on Unified Communications Manager, navigate to Cisco Unified CM Administration configuration pages, select Devices drop down list and from there pick up Phones and click on Find/List. You should receive output that indicates that SCCP phone is registered, as follows:

Please notice that in upper right corner on your phone you will be able to see your configured directory number followed by the SCCP profile name (in our example it is (1003)CallManager), and also, registration status will be indicated by the small VoIP icon in bottom part of the screen of your Nokia Eseries phone. Once you have your profile registered with Unified Communications Manager, you can start making VoIP phone calls and you can start using productivity features that we already mentioned in previous text.

For more details about Call Connect please refer to following links:

1. Nokia Call Connect For Cisco
2. Nokia Call Connect For Cisco: Licensing and Support

During last few months I was intensively working with Cisco Unified Communications Manager, previously called just Call Manager in order to obtain Cisco Unified Comminications IP Telephony (CIPT) certification. Since I am coming from the ‘voice’ field when I started to work with Cisco products it was logical to me to check what is Cisco offering in that field. And I remember that I was impressed. Six different certifications after CCNA and two possible CCVP paths. Plenty of different solutions, gateways, protocols and such was enough challenging to me. First thing that I needed is to make clear decision of which CCVP path to follow. One is covering CallManager (version 6.X called CUCM) in two parts (CIPT1 and CIPT2) and other, old one, is covering CallManager (Cisco Unified CallManager 4.X) throught one certification mixing everything with additional Gateway/Gatekeeper certification and that path is about to reach end of life on December, 31. Since I was working with web based call processing device in past I decided to go with actual Unified Communications Manager CCVP path (because CUCM is web based call processing solution as well). One of the exams on that path is Cisco Unified Communications IP Telephony Part 1 which I have passed today.  There was 60 questions and you needed to score around 80% to pass it. There are single choice, multiple choice and drag and drop questions. It was not that easy at all although I was preparing for it for few months and that is normal because this is very complex solution. When it comes to CUCM I need to say that I was quite surprised of number of features that it can provide. It is very powerful, high available and redundant call processing solution which is covering advanced mobility, call coverage and other solutions in very organized, logical and intelligent way. I was preparing my certification following multiple documentation sources and by following quick reference. Note that CallManager can be installed in VMware which is a good thing, because when it comes to practicing you won’t spend lot of money to build complete testing environment. Also, please note that there are many good CCVP blogs which can help a lot and I will post few links below the text. If you need some additional information on CallManager or this certification, feel free to contact me, I will be willing to help.

1. Cisco Unified Communications Manager
2. CCVP certification paths
3. CCIE12932 blog
4. Chris’ CCVP blog

One of the best Cisco Unified Communications Manager VoIP features is Extension Mobility in my personal opinion. It allows you to temporarily configure another IP Phone as your own by logging in to that phone. Once logged in you will have your number, speed dials and etc. onto that phone, and if you are working as teleworker you would know to appreciate those kind of options. Following text describes how to configure CallManager 6.0 to support Extension Mobility.

Task 1: Verify Extension Mobility Service is Running

Step 1: From the Navigation menu select Cisco Unified CallManager Serviceability

Step 2: Select Tools>Control Center – Feature Services

Step 3: Make sure that the Cisco Extension Mobility service shows status Activated

Task 2: Configure Extension Mobility Service

Step 1: From the Navigation menu select Cisco Unified CallManager Administration

Step 2: Select Device>Device Settings>Phone Services

Step 3: Click Add New

Step 4: In the Service Name field, type Extension Mobility
Step 5: In the Service Description field, type Login and logout service
Step 6: In the Service URL field, Enter the following URL: http://YOURCUCMIPADDRESS/emapp/EMAppServlet?device=#DEVICENAME#

Step 7: Click Save

Task 3: Modify Enterprise Parameters to Reflect IP Address of CallManager (remove DNS reliance)

Step 1: Select System>Enterprise Parameters

Step 2: Under Phone URL parameters, change all fields to reflect IP addresses instead of hostnames. Change ONLY the host name, not the reset of the field.

Step 3: Click Save
Step 4: Click Ok from the pop-up warning.
Step 5: Click Reset
Step 6: In the pop-up window select Reset
Step 7: Click Close

Task 4: Create Device Profile Default for Each Phone Model that shall Support Cisco Extension Mobility (this step is optional)

Step 1: Select Device>Device Settings>Default Device Profile
Step 2: From the drop down list, select the phone model to be configured, for example, Cisco 7960.
Step 3: Under Description, enter a description of this profile.
Step 4: Under Phone Button Template, select Standard 7960 SCCP.
Step 5: Click Save
Step 6: Repeat for each model phone to be configured

Task 5: Create Device User Profile for a User

Step 1:  Choose Device>Device Settings>Device Profile and click Add New.

Step 2: From the drop down list, select the phone model to be configured, for example, Cisco 7960
Step 3: Click Next
Step 4: Enter a Device Profile Name (in this example KemalSanjtaProfile).
Step 5: From the Phone Button Template field, select Standard 7960 SCCP.
Step 6: Click Save.

Step 7: On the left hand side of the screen, click the link Line [1] – Add a new DN.

Step 8: Choose a valid DN from your NIP, enter that DN in the Directory Number field.
Step 9: Under Route Partition, select your city’s Headquarters Partition.

Step 10: Under Directory Number Settings choose a CSS of appropriate access.

Step 11:  Enter any Call Forward and Call Pickup Settings as necessary.
Step 12: In the Display (Internal Caller ID)
Step 13: Click Save.
Step 14: From the Related Links: menu, select Subscribe/Unsubscribe Services.

Step 15: In the Select a Service, select Extension Mobility, then click Next.

Step 16: Click Subscribe.

Step 17: Click Save.
Step 18: Repeat steps 7-13 for any additional lines.

Task 6: Associate User Device Profile to a User

Step 1: From the menu, select User Management>End User.

Step 2: Click Find
Step 3: Select the user from the list that matches the profile that was created.

Step 4: Under Extension Mobility>Available Profiles, select the profile that was created in the previous exercise and move it to the Controlled Profiles selection (in our example it is KemalSanjtaProfile).

Step 5: Under Default Profile, select the profile.
Step 6: Click Save.

Task 7: Configure and Subscribe Cisco Unified Ip Phones to Service and Enable it.

Step 1:  Select Device>Phone from the menu.

Step 2:  Select the phone from the list of devices.

Step 3: In the Related Links: field, select Subscribe/Unsubscribe Services and click Go

Step 4: In the pop-up window, under Service Information, in the Select a Service pull down menu, select Extension Mobility.

Step 5: Click Next
Step 6: Click Subscribe

Step 7: Click Save

Step 8: Close the pop-up window

Step 9: Under Extension Information , check the Enable Extension Mobility box.
Step 10: Under the Logout Profile field, select – Use Current Device Settings –
Step 11: Click Save.

Step 12: Click Ok from the pop-up warning.
Step 13: Click Reset
Step 14: In the pop-up window select Reset.
Step 15: Click Close.

Note: This post has been updated on 12/03/2010 in order to describe how to assign Extension Mobility Phone service to Device Profile (including screenshots).

1. Validate if a slot on AS5350 are free with “sh chassis slot” command

example:
hostname#sh chassis slot

Slot 1:
DFC type is AS5350 NP60 DFC

OIR events:

DFC State is DFC_S_OPERATIONAL

Slot 2:
DFC type is AS5350 Empty DFC
DFC is not powered

OIR events:

Slot 3:
DFC type is AS5350 T1 2 PRI DFC

OIR events:

DFC State is DFC_S_OPERATIONAL

2. If slot 2 are available, do a “busyout 2″ (in enable mode) to deactivate correctly the slot no 2.

3. Insert the QUAD into the slot 2 carefully

4. Wait 10 seconds et validate the new QUAD aren’t in progress “show busyout 2″

example (You should see something similar).:
hostname#sh busyout 2
Busyout status for trunk DFC slot = 2:
(p – pending, s – static(cfg/exec), d – dynamic, n – none)

2/0 : n n n n n n n n n n n n n n n n n n n n n n n n
2/1 : n n n n n n n n n n n n n n n n n n n n n n n n
2/2 : n n n n n n n n n n n n n n n n n n n n n n n n
2/3 : n n n n n n n n n n n n n n n n n n n n n n n n
hostname#

5. Apply these settings to create new controller T1 for new QUAND on slot 2

controller T1 2/0
framing esf
linecode b8zs
pri-group timeslots 1-24 nfas_d primary nfas_int 0 nfas_group 2
shutdown
!
controller T1 2/1
framing esf
linecode b8zs
pri-group timeslots 1-24 nfas_d backup nfas_int 1 nfas_group 2
shutdown
!
controller T1 2/2
framing esf
linecode b8zs
pri-group timeslots 1-24 nfas_d primary nfas_int 0 nfas_group 3
shutdown
!
controller T1 2/3
framing esf
linecode b8zs
pri-group timeslots 1-24 nfas_d backup nfas_int 1 nfas_group 3
shutdown
!

6. Create new voice-port for D-channel

voice-port 2/0:D
bearer-cap Speech
!
voice-port 2/2:D
bearer-cap Speech
!

7. Configure the D channel settings

interface Serial2/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice modem
no cdp enable
!
interface Serial2/2:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice modem
no cdp enable
!

8. Assosiate the incoming regional number with the bearer

dial-peer voice 300 pots
incoming called-number 517…….
direct-inward-dial
port 2/0:D
!
dial-peer voice 310 pots
incoming called-number 457…….
direct-inward-dial
port 2/0:D
!
dial-peer voice 320 pots
incoming called-number 817…….
direct-inward-dial
port 2/0:D
!
dial-peer voice 330 pots
incoming called-number 417…….
direct-inward-dial
port 2/0:D
!
dial-peer voice 340 pots
incoming called-number 437…….
direct-inward-dial
port 2/0:D
!

9. Activate the new controller T1 (“no shutdown”)

10. Validate that controller T1 came UP “sh isdn status”; “sh isdn service”

11. copy run start

More than six months from getting my CCNA certificate, I have passed CCNA Voice (few weeks ago actually, but I am not refreshing this blog as much as I would like to). For me, that was logical step to take, because I was working few years in VoIP industry and I was interested in Cisco’s way of solving some VoIP based tasks, like voice routing, productivity features (music on hold, call transfer, blind and consultative, after hours call blocking, directory, call forwarding, call park/pickup and so on). CCNA Voice cert is covering all of those topics in details including setup of Cisco Unity (their voicemail solution), codecs and many other configuration based things that you could face as real-world requirements (like PSTN fail-over for example). There is up to 65 questions on the test and you are having two hours for that. There is just one simulation on the exam (I have expected more, but there is no as much as on the CCNA exam). Questions are in the form of the single answer, multiple choice and drag and drop. It was real pleasure for me to prepare this exam, since I was having two deployments of Cisco VoIP in prior to my decision to get certified in this field, and at the moment I am dealing with Cisco voice gateways. In next few months I will try to get some more voice certs, depending on my free time. Everyone interested in voice over ip, or generally in voice and is Cisco oriented should check this huge and interesting area.

Few months ago i decided to buy Nokia E71. Although i was not fan of those full qwerty mobiles (they always seemed to be so big) i have decided to buy this one after realising that it is piece of good and really powerful hardware (400 Mhz ARM processor, almost like my first PC). And really, i found it to be really good. From connectivity point of view we are dealing with mobile phone that is working with Bluetooth, Wireless LAN and SIP (which i personally admire as killer feature) VPN and other interesting stuff, which is more then enough for normal business user. As i already mentioned, the best thing for me that it is working with SIP. In company that I am working for, we have Asterisk telephony solution which is working with SIP and IAX protocol so i have my personal extension for mobile phone and when i am working out of office, in area that is covered with wireless internet connectivity, i can make free phone calls (at the moment we are considering one of those nice analog cards from Digium or Sangoma so that we can make calls to PSTN) which is really good. This is classical business oriented phone with a lot of nice features that could be used in different environments and for different business needs. Also, it is working with Simbian so there is a lot of good and useful applications available. I would fully recommend this phone to anyone who needs business phone.

1. Full phone specifications

…the Qtopia framework/stack has been used in about 40 VoIP devices, making it the “dominant Linux development platform for VoIP/WiFi devices,” according to Trolltech

Interesantan ?lanak na temu možete pro?itati na ovom linku.