Please CLICK below ADD to make prosper this BLOG

Thursday, February 28, 2013

Position Vacant - Assistant Manager – Networks & Systems


Position Vacant
Assistant Manager – Networks & Systems

 

Position Title
Assistant Manager – Networks & Systems
Department
Information Technology
Reporting Relationship
Senior Strategist, IT

 

Scope of the Position
Assistant Manager – Network & Systems Administration is responsible for effective provisioning, installation/configuration, operation, and maintenance of systems hardware, software and related infrastructure. This individual participates in technical research and development to enable continuing innovation within the infrastructure. This individual ensures that system hardware, operating systems, software systems, and related procedures adhere to organizational values, enabling staff, faculty, and Partners. This individual will assist project teams with technical issues in the Initiation and Planning phases of our standard Project Management Methodology. These activities include the definition of needs, benefits, and technical strategy; research & development within the project life-cycle; technical analysis and design; and support of operations staff in executing, testing and rolling-out the solutions. This individual is accountable for the following systems: Linux and Windows systems that support IS infrastructure; Enterprise Applications Campus Management; Database Administration on SQL, MYSQL and Oracle on these systems include SA engineering including provisioning, operations, support, maintenance and research and development to ensure continual innovation.

 

Summary of Key Functions
Characteristics, Duties and Responsibilities:
Development:
  • Responsible to work with consultants and vendor for requirements gathering, design, configuration, technical specifications, integration, data migration, testing, documentation, training, pre & post implementation setups / configuration and support.
  • Install new / rebuild existing servers and configure hardware, peripherals, services, settings, directories, storage, etc. in accordance with standards and project/operational requirements.
  • Research and recommend innovative, and where possible automated approaches for system administration tasks. Identify approaches that leverage our resources and provide economies of scale.
  • Perform daily system monitoring, verifying the integrity and availability of all hardware, server resources, systems and key processes, reviewing system and application logs, and verifying completion of scheduled jobs such as backups.
  • Perform regular security monitoring to identify any possible intrusions.
  • Provide Tier III/other support per request from various constituencies. Investigate and troubleshoot issues.
  • Repair and recover from hardware or software failures. Coordinate and communicate with impacted constituencies.
  • Apply OS patches and upgrades on a regular basis, and upgrade administrative tools and utilities. Configure / add new services as necessary. Maintain operational, configuration, or other procedures.
  • Perform periodic performance reporting to support capacity planning. Perform ongoing performance tuning, hardware upgrades, and resource optimization as required. Configure CPU, memory, and disk partitions as required. Maintain data center environmental and monitoring equipment.
  • Functions as a lead worker doing the work similar to those in the work unit; responsibility for training, instruction, setting the work pace, and possibly evaluating performance.
  • Establishing the needs of users and monitoring user access and security; monitoring performance and managing parameters to provide fast query responses to front-end users.
  • Mapping out the conceptual design for a planned database in outline; considering both back-end organization of data and front-end accessibility for end-users.
  • Refining the logical design so that it can be translated into a specific data model; further refining the physical design to meet system storage requirements.
  • Installing and testing new versions of the DBMS; maintaining data standards, including adherence to the Data Protection Act.
  • Writing systems documentation, including data standards, procedures and definitions for the data dictionary (metadata); controlling access permissions and privileges.
  • Developing, managing and testing back-up and recovery plans; ensuring that storage, archiving, back-up and recovery procedures are functioning correctly.
  • Capacity planning; working closely with IT project managers, database programmers and multimedia programmers.
  • Communicating regularly with technical, applications and operational staff to ensure database integrity and security.
  • Commissioning and installing new applications and customizing existing applications in order to make them fit for purpose.

 

RequiredQualification, Experience and Skills
The Assistant Manager – Networks & Systems will have:.
  • Bachelor’s degree in Computer Science, Information Technology, Information Systems, Computer Engineering or equivalent combination of experience, education and training from a reputed university.
  • Overall 03 years’ System Administration experience ideally at Higher Education institution, development / software house or at Systems Integrators SI’s and vendor business.
  • Proven track record of managing / assisting on Networking projects, designing and configuring wired / wireless networks is must.
  • Experience in designing and configuring telecommunication systems project is an advantage.
  • Systems Administration/System Engineer certification in Linux and Microsoft.
  • Systems Administration/System Engineer certification in Microsoft Exchange.
  • Systems Administration/System Developer certification in Microsoft Lync Servers.
  • Systems Administration experience in MS SharePoint and other open source systems.
  • Application deployments of Open Source platforms, its customization and integrations.
  • Good knowledge of ASP.NET, PHP, Java and open source platforms.
  • Good knowledge of Web Applications deployment and integration.
  • Excellent communication skills and the ability to work with stakeholders of differing levels.
  • Attention to detail and a flexible approach to work.
  • Excellent planning, coordination and prioritization skills and proven capacity to undertake varied tasks simultaneously within stringent deadlines.
  • Experience of working at jobs requiring close attention to detail.
  • Proven record of teamwork and project management in previous roles.
  • Self-motivated and able to meet agreed objectives on own initiative.

 

Habib Universtiy
Habib University will be a pioneering institution providing a rich liberal arts education to the youth of Pakistan to create a generation of socially responsible and critically conscious individuals, who can bring Pakistan to the forefront not only economically and financially, but intellectually as well. The University is being established in Karachi, Pakistan. Habib University will initiate its classes in the year 2014 with 200 students, offering four undergraduate programs offered through the School of Arts, Humanities and Social Sciences and the School of Science and Engineering. The University will offer a rich, interdisciplinary learning fabric by bringing together the innovative spirit of Entrepreneurship with high quality Science, Engineering and Liberal Arts education. The University will be a modern learning space fully equipped with state-of-the-art research and teaching facilities in order to attract the best faculty from around the world. For more information please visit: http://habib.edu.pk/

 

About Habib Universtiy Foundation
Habib University Foundation (H.U.F.) is a not-for-profit organization, which commenced its work in 2007. H.U.F. responds to existing gaps in the educational scenario of Pakistan, by supporting educational initiatives, research and innovation. It is focused on improving the status of education within the country by supporting research, planning and implementation of innovative educational models. H.U.F. extends support at all tiers of education delivery within the country including higher education, vocational skill development, pre-tertiary education and research and advocacy. For more information on the foundation please visit: http://www.huf.org.pk

 

About House of Habib
The House of Habib (HoH) is a well-established industrial house in Pakistan. It is equally famous for its philanthropic contributions towards the development of Pakistani society. HoH has a rich tradition in commerce and finance that dates all the way back to 1841 when Esmail Ali of Jamnagar, India, set up a utensil factory in Bombay (present day Mumbai). Since then, his descendants have built on his hard work and today the House of Habib has business interests across varied industries. Today, the House of Habib’s portfolio boasts of some of the most successful and financially stable companies in Pakistan including Indus Motors, AuVitronics Limited, DYNEA Pakistan Limited, Shabbir Tiles and Ceramics Limited, Thal Limited, Habib Bank AG Zurich etc.We are an equal opportunity employer and appreciate diversity in our organization. We assure you that at the foundation, you will interact with people from various backgrounds, increasing your exposure to various skill sets and opportunities.

 

Offered Benefits
The foundation offers a comprehensive salary package that is designed to meet the needs of a diverse workforce. In addition to this, employees are also entitled to become members of the Foundation’s Provident Fund, Group Life Insurance and Hospitalization/Health Coverage schemes. We also support personal and professional development of our employees and offer excellent professional learning and development possibilities. .

 

Our Culture
The foundation firmly believes that trust and respect for the potential of people is the key to creating an inspiring and exciting work environment which promotes creativity and innovation, and a sense of self accountability and pro activeness. Our values guide the work we do and help us create a unique culture where each of our colleagues feel valued and empowered. .

 

APPLY AT:

 

http://www.huf.org.pk/employment/ADIT.htm?goback=.gde_4267483_member_217974687

comparasion Cisco vs any product

 


Comparison between Two core switches  HP 8400 & 5200 series core switch & Cisco Core switch Cataylst 4500 series core switch. You can understand which core switch is better for u. Cost also a major factor cisco switch cost is high and HP core switch low but performance you can make your decision which is more reliable to you at your environment.

Wednesday, February 27, 2013

How to configure Cisco Callmanager Express with 3rd Party Device

 


Call Manager Express basically available at ISR [Integrated service router because it has minimum features for all flavours like IPS, IDS, VPN, Routing and etc.

Had to configure a Cisco Callmanager Express to accept connections from 3rd party SIP phones via the Internet. The SIP phones need to reach each other, their voicemail and PSTN phones via ISDN breakout.
Keep in mind that, as always, public IP's have been changed to private ones. Phone numbers are also fake. Signalling & RTP communication is NOT encrypted in this example! Be aware of that! You could tunnel this traffic through a VPN. Registration would then work too & everything should be fine (except for the additional delays ...). ISDN configuration is for German PSTN, but you should be able to modify it for your needs. If you have trouble to get this to work, try this debug commands:

debug ccsip all (Be carefull, some phones fire off over 20 register requests per second. This usually only happens if the phone is not able to register, but it might freeze your router. This is not a theoretically option!)
debug voice register errors
debug voice register events

Check your systems firewall settings if you use software phones. Might be a good idea to deactivate it temporarily for verifying functionality.

At first I would try to use the X-Lite client. That's a client that usually always works first. Most tolerant one for NAT issues. In general almost any third party SIP client, even IPhones, should work in this implementation. Before delivering such a solution you should always verify functionality thoroughly. Some problems arise after a longer period of time because of timeouts etc..
This is only a short abstract. If you have any suggestions or coments - feel free to post them.

Overview

Configuration of Cisco Callmanager Express


sipgateway#sh run
Building configuration...


Current configuration : 8775 bytes
!
version 15.1
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname sipgateway
!
boot-start-marker
boot system flash:c2900-universalk9-mz.SPA.151-4.M1.bin
boot-end-marker
!
!
logging buffered 100000
!
aaa new-model
!
!
aaa authentication login default local
aaa authorization exec default local
!
!
!
!
!
aaa session-id common
clock timezone MEZ 1 0
clock summer-time MESZ recurring last Sun Mar 2:00 last Sun Oct 3:00
network-clock-participate wic 0
network-clock-select 1 BRI0/0/1
!
no ipv6 cef
no ip source-route
ip cef
!
!
!
!
!
ip domain name lab.local
ip name-server 172.20.21.5
multilink bundle-name authenticated
!
!
!
!
isdn switch-type basic-net3
!
voice-card 0
 dsp services dspfarm
!
!
voice call disc-pi-off
!
voice service voip
 allow-connections sip to sip
 redirect ip2ip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0
  registrar server expires max 600 min 60
  no call service stop
!
voice class codec 10
 codec preference 1 g711ulaw
!
!
voice register global
 mode cme
 source-address 172.20.21.165 port 5060
 max-dn 35
 max-pool 10
 authenticate register ==> This is needed, because phones are not localy connected.
 authenticate realm lab.local ==> This is needed by some SIP phones to switch to digest auth.
 timezone 21
 time-format 24
 date-format D/M/Y
 voicemail 88888888
 tftp-path flash:
 create profile sync 0429414478545137
!
voice register dn  1
 number 12344887
 call-forward b2bua unregistered 88888888 
 allow watch
 name Test1
 label 12344887
 mwi
!
voice register dn  2
 number 12344898
 allow watch
 name Test2
 label 12344898
 mwi
!
voice register dn  4
 number 12344971
 call-forward b2bua unregistered 88888888 
 allow watch
 name Test4
 label 12344971
 mwi
!
voice register dn  5
 number 12341453
 allow watch
 name Test5
 label 12341453
 mwi
!
voice register dn  7
 number 12341455
 allow watch
 name Test7
 label 12341455
 mwi
!
voice register pool  1
 id mac 0000.0000.0000 ==> Mac is irrelevant. Auth is now digest based.
 number 1 dn 1
 presence call-list
 dtmf-relay rtp-nte
 username 12344887 password 1234
 codec g711ulaw
!
voice register pool  2
 id mac 0000.0000.0000
 number 1 dn 2
 presence call-list
 dtmf-relay rtp-nte
 username 12344898 password 1234
 codec g711ulaw
!
voice register pool  4
 id mac 0000.0000.0000
 number 1 dn 4
 presence call-list
 dtmf-relay rtp-nte
 username 12344971 password 1234
 codec g711ulaw
!
voice register pool  5
 id mac 0000.0000.0000
 number 1 dn 5
 presence call-list
 dtmf-relay rtp-nte
 username 12341453 password 1234
 codec g711ulaw
!
voice register pool  7
 id mac 0000.0000.0000
 number 1 dn 7
 presence call-list
 dtmf-relay sip-notify
 username 12341455 password 1234
 codec g711ulaw
!
!
!
voice translation-rule 5
 rule 1 /^\(.*\)/ /30\1/ type any national
!
voice translation-rule 10
 rule 1 /^\(.*\)/ /0\1/ type subscriber unknown
 rule 2 /^\(.*\)/ /00\1/ type national unknown
 rule 3 /^\(.*\)/ /000\1/ type international unknown
!
!
voice translation-profile From-PSTN
 translate calling 10
!
voice translation-profile To-PSTN
 translate calling 5
!
!
license udi pid CISCO2901/K9 sn 12341234
license accept end user agreement
hw-module ism 0
!
hw-module pvdm 0/0
!
!
!
username labtest privilege 15 labt3st
!
redundancy
!
!
!
interface Loopback0
 ip address 172.20.20.1 255.255.255.252
!
interface Embedded-Service-Engine0/0
 no ip address
 shutdown
!
interface GigabitEthernet0/0
 description LAN Interface
 ip address 172.20.21.165 255.255.255.248
 duplex auto
 speed auto
!
interface ISM0/0
 ip unnumbered Loopback0
 service-module ip address 172.20.20.2 255.255.255.252
 !Application: CUE Running on ISM
 service-module ip default-gateway 172.20.20.1
!
interface GigabitEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface ISM0/1
 description Internal switch interface connected to Internal Service Module
 no ip address
 shutdown
!
interface BRI0/0/0
 no ip address
 shutdown
 isdn switch-type basic-net3
 isdn point-to-point-setup
 isdn incoming-voice voice
!
interface BRI0/0/1
 no ip address
 isdn switch-type basic-net3
 isdn point-to-point-setup
 isdn incoming-voice voice
!
ip http server
ip http access-class 24
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:
!
ip route 0.0.0.0 0.0.0.0 172.20.21.161
ip route 172.20.20.2 255.255.255.255 ISM0/0
!
!
!
!
!
control-plane
!
!
voice-port 0/0/0
 compand-type a-law
 cptone DE
 bearer-cap Speech
!
voice-port 0/0/1
 compand-type a-law
 cptone DE
 bearer-cap Speech
!
!
dial-peer voice 1 pots
 description ISDN
 translation-profile incoming From-PSTN
 translation-profile outgoing To-PSTN
 destination-pattern 0.T
 incoming called-number .
 direct-inward-dial
 port 0/0/1
!
dial-peer voice 5 voip
 destination-pattern 88888888
 session protocol sipv2
 session target ipv4:172.20.20.2
 incoming called-number .
 voice-class codec 10 
 dtmf-relay sip-notify
 no vad
!
!
gateway
 timer receive-rtp 1200
!
sip-ua
!
end

sipgateway#        

Verify registration

sipgateway#sh sip-ua status registrar
Line          destination      expires(sec)  contact
transport     call-id
              peer
============================================================
12341455      172.20.22.52     597           172.20.22.52
UDP           g7ngEr-P2hu1kPJ6mDgWP8FNWrPJDIql             
              40002

These are the phone configs I tested:


Android CSipSimple Settings
 Accountname: 12344971
 Send own number: 12344971
 SIP Server: 172.20.21.165
 Username: 12344971
 Password: 1234
 Proxy: 172.20.21.165


Phoner Lite Settings
 Configuration -> Server
  Proxy/registrar: 172.20.21.165
  STUN Server: stun.counterpath.com
  Domain/Realm: 172.20.21.165
  Check Registration
 Configuration -> User
  Username: 12341453
  Shown username: 12341453
  Password: 1234
  Authentication name: 12341453
  Number: 12341453
 Configuration -> Network
  Check preferred connection type: UDP
  Check Windows Firewall

Xlite (ver 4.0) settings
 Softphone -> Account Settings -> Account
  Check allow this account for call
  User ID: 12341453
  Domain: 172.20.21.165
  Password: 1234
  Authorization name: 12341453
  Check Domain Proxy to register with Domain and receive calls
  Check outbound via domain
 Softphone -> Account Settings -> Topology
  Autodetect firewall traversal method using ICE
 Softphone -> Preferences -> Advanced
  Check send DTMF via RFC2833

Snom 360 Settings
 Identity1
  Login
   Account: 12344887
   Password: 1234
   Registrar: 172.20.21.165
   Authentication Username: 12344887
  SIP
   Check Support broken Registrar
  NAT
   Check Offer ICE
   STUN Server: stun.counterpath.com

How to configure

Had to configure a Cisco Callmanager Express to accept connections from 3rd party SIP phones via the Internet. The SIP phones need to reach each other, their voicemail and PSTN phones via ISDN breakout.
Keep in mind that, as always, public IP's have been changed to private ones. Phone numbers are also fake. Signalling & RTP communication is NOT encrypted in this example! Be aware of that! You could tunnel this traffic through a VPN. Registration would then work too & everything should be fine (except for the additional delays ...). ISDN configuration is for German PSTN, but you should be able to modify it for your needs. If you have trouble to get this to work, try this debug commands:

debug ccsip all (Be carefull, some phones fire off over 20 register requests per second. This usually only happens if the phone is not able to register, but it might freeze your router. This is not a theoretically option!)
debug voice register errors
debug voice register events

Check your systems firewall settings if you use software phones. Might be a good idea to deactivate it temporarily for verifying functionality.

At first I would try to use the X-Lite client. That's a client that usually always works first. Most tolerant one for NAT issues. In general almost any third party SIP client, even IPhones, should work in this implementation. Before delivering such a solution you should always verify functionality thoroughly. Some problems arise after a longer period of time because of timeouts etc..
This is only a short abstract. If you have any suggestions or coments - feel free to post them.

Overview

Configuration of Cisco Callmanager Express


sipgateway#sh run
Building configuration...


Current configuration : 8775 bytes
!
version 15.1
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname sipgateway
!
boot-start-marker
boot system flash:c2900-universalk9-mz.SPA.151-4.M1.bin
boot-end-marker
!
!
logging buffered 100000
!
aaa new-model
!
!
aaa authentication login default local
aaa authorization exec default local
!
!
!
!
!
aaa session-id common
clock timezone MEZ 1 0
clock summer-time MESZ recurring last Sun Mar 2:00 last Sun Oct 3:00
network-clock-participate wic 0
network-clock-select 1 BRI0/0/1
!
no ipv6 cef
no ip source-route
ip cef
!
!
!
!
!
ip domain name lab.local
ip name-server 172.20.21.5
multilink bundle-name authenticated
!
!
!
!
isdn switch-type basic-net3
!
voice-card 0
 dsp services dspfarm
!
!
voice call disc-pi-off
!
voice service voip
 allow-connections sip to sip
 redirect ip2ip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0
  registrar server expires max 600 min 60
  no call service stop
!
voice class codec 10
 codec preference 1 g711ulaw
!
!
voice register global
 mode cme
 source-address 172.20.21.165 port 5060
 max-dn 35
 max-pool 10
 authenticate register ==> This is needed, because phones are not localy connected.
 authenticate realm lab.local ==> This is needed by some SIP phones to switch to digest auth.
 timezone 21
 time-format 24
 date-format D/M/Y
 voicemail 88888888
 tftp-path flash:
 create profile sync 0429414478545137
!
voice register dn  1
 number 12344887
 call-forward b2bua unregistered 88888888 
 allow watch
 name Test1
 label 12344887
 mwi
!
voice register dn  2
 number 12344898
 allow watch
 name Test2
 label 12344898
 mwi
!
voice register dn  4
 number 12344971
 call-forward b2bua unregistered 88888888 
 allow watch
 name Test4
 label 12344971
 mwi
!
voice register dn  5
 number 12341453
 allow watch
 name Test5
 label 12341453
 mwi
!
voice register dn  7
 number 12341455
 allow watch
 name Test7
 label 12341455
 mwi
!
voice register pool  1
 id mac 0000.0000.0000 ==> Mac is irrelevant. Auth is now digest based.
 number 1 dn 1
 presence call-list
 dtmf-relay rtp-nte
 username 12344887 password 1234
 codec g711ulaw
!
voice register pool  2
 id mac 0000.0000.0000
 number 1 dn 2
 presence call-list
 dtmf-relay rtp-nte
 username 12344898 password 1234
 codec g711ulaw
!
voice register pool  4
 id mac 0000.0000.0000
 number 1 dn 4
 presence call-list
 dtmf-relay rtp-nte
 username 12344971 password 1234
 codec g711ulaw
!
voice register pool  5
 id mac 0000.0000.0000
 number 1 dn 5
 presence call-list
 dtmf-relay rtp-nte
 username 12341453 password 1234
 codec g711ulaw
!
voice register pool  7
 id mac 0000.0000.0000
 number 1 dn 7
 presence call-list
 dtmf-relay sip-notify
 username 12341455 password 1234
 codec g711ulaw
!
!
!
voice translation-rule 5
 rule 1 /^\(.*\)/ /30\1/ type any national
!
voice translation-rule 10
 rule 1 /^\(.*\)/ /0\1/ type subscriber unknown
 rule 2 /^\(.*\)/ /00\1/ type national unknown
 rule 3 /^\(.*\)/ /000\1/ type international unknown
!
!
voice translation-profile From-PSTN
 translate calling 10
!
voice translation-profile To-PSTN
 translate calling 5
!
!
license udi pid CISCO2901/K9 sn 12341234
license accept end user agreement
hw-module ism 0
!
hw-module pvdm 0/0
!
!
!
username labtest privilege 15 labt3st
!
redundancy
!
!
!
interface Loopback0
 ip address 172.20.20.1 255.255.255.252
!
interface Embedded-Service-Engine0/0
 no ip address
 shutdown
!
interface GigabitEthernet0/0
 description LAN Interface
 ip address 172.20.21.165 255.255.255.248
 duplex auto
 speed auto
!
interface ISM0/0
 ip unnumbered Loopback0
 service-module ip address 172.20.20.2 255.255.255.252
 !Application: CUE Running on ISM
 service-module ip default-gateway 172.20.20.1
!
interface GigabitEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface ISM0/1
 description Internal switch interface connected to Internal Service Module
 no ip address
 shutdown
!
interface BRI0/0/0
 no ip address
 shutdown
 isdn switch-type basic-net3
 isdn point-to-point-setup
 isdn incoming-voice voice
!
interface BRI0/0/1
 no ip address
 isdn switch-type basic-net3
 isdn point-to-point-setup
 isdn incoming-voice voice
!
ip http server
ip http access-class 24
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:
!
ip route 0.0.0.0 0.0.0.0 172.20.21.161
ip route 172.20.20.2 255.255.255.255 ISM0/0
!
!
!
!
!
control-plane
!
!
voice-port 0/0/0
 compand-type a-law
 cptone DE
 bearer-cap Speech
!
voice-port 0/0/1
 compand-type a-law
 cptone DE
 bearer-cap Speech
!
!
dial-peer voice 1 pots
 description ISDN
 translation-profile incoming From-PSTN
 translation-profile outgoing To-PSTN
 destination-pattern 0.T
 incoming called-number .
 direct-inward-dial
 port 0/0/1
!
dial-peer voice 5 voip
 destination-pattern 88888888
 session protocol sipv2
 session target ipv4:172.20.20.2
 incoming called-number .
 voice-class codec 10 
 dtmf-relay sip-notify
 no vad
!
!
gateway
 timer receive-rtp 1200
!
sip-ua
!
end

sipgateway#        

Verify registration

sipgateway#sh sip-ua status registrar
Line          destination      expires(sec)  contact
transport     call-id
              peer
============================================================
12341455      172.20.22.52     597           172.20.22.52
UDP           g7ngEr-P2hu1kPJ6mDgWP8FNWrPJDIql             
              40002

These are the phone configs I tested:


Android CSipSimple Settings
 Accountname: 12344971
 Send own number: 12344971
 SIP Server: 172.20.21.165
 Username: 12344971
 Password: 1234
 Proxy: 172.20.21.165


Phoner Lite Settings
 Configuration -> Server
  Proxy/registrar: 172.20.21.165
  STUN Server: stun.counterpath.com
  Domain/Realm: 172.20.21.165
  Check Registration
 Configuration -> User
  Username: 12341453
  Shown username: 12341453
  Password: 1234
  Authentication name: 12341453
  Number: 12341453
 Configuration -> Network
  Check preferred connection type: UDP
  Check Windows Firewall

Xlite (ver 4.0) settings
 Softphone -> Account Settings -> Account
  Check allow this account for call
  User ID: 12341453
  Domain: 172.20.21.165
  Password: 1234
  Authorization name: 12341453
  Check Domain Proxy to register with Domain and receive calls
  Check outbound via domain
 Softphone -> Account Settings -> Topology
  Autodetect firewall traversal method using ICE
 Softphone -> Preferences -> Advanced
  Check send DTMF via RFC2833

Snom 360 Settings
 Identity1
  Login
   Account: 12344887
   Password: 1234
   Registrar: 172.20.21.165
   Authentication Username: 12344887
  SIP
   Check Support broken Registrar
  NAT
   Check Offer ICE
   STUN Server: stun.counterpath.com

Tuesday, February 26, 2013

CISCO IOS version 15.0

Developed for wide deployment in the world’s most demanding enterprise, access, and service provider aggregation networks, Cisco IOS Software Release 15 M and T provides a comprehensive portfolio of Cisco technologies, including the leading-edge functionality and hardware support from Releases 12.4 and 12.4T.

Release 15 M and T key innovations span multiple technology areas, including Security, Voice, High Availability, IP Routing and Multicast, Quality of Service (QoS), IP Mobility, Multiprotocol Label Switching (MPLS), VPNs, and Embedded Management.
Release 15 M and T provides customers new feature release delivery and hardware support in a shorter amount of time, broadened feature consistency, more reliable new feature release and rebuild schedules, proactive release support lifecycle policies, and easier software selection, deployment, and migration guidelines.

15 M (Extended Maintenance) releases delivered on a more frequent basis (approximately every 16 months), enabling customers to qualify, deploy, and remain on a release longer with active support. Release 15.0(1)M (FCS November, 2009) is the first 15 M release.

15 T (Standard Maintenance) releases (in between 15 M releases) ideal for the very latest features and hardware support before the next 15 M release becomes available. Each 15 T release receives bug-fix rebuild support for 13 months, plus an additional 6 months support for security/vulnerability issues such as Product Security Incident Response Team (PSIRT) advisories. Release 15.1(1)T (FCS March, 2010) is the first 15 T release.

Release numbering schemes for release 15M&T:


Maintenance rebuild methodology for release 15M&T:


Cisco IOS Software Release 15M&T Lifecycle:

MilestoneDefinitionDate
First Customer Shipment (FCS)The date the software release is first available to customers on Cisco Software Download CenterDay 0
Maintenance Deployment Point (MD Point)Maintenance Deployment (MD) designated M Releases are widely deployable, as proven through extensive system testing, which continues as part of Cisco’s ongoing commitment to improve and increase reliability and quality. A set of stringent criteria is used to analyze the quality of the release, such as high customer satisfaction, expansive customer deployment, demonstrated software reliability, and rigorous ongoing testing.Typically 9 to 12 months (3rd or 4th maintenance rebuild) after the initial posting (FCS date) of the release on Cisco.com Software Download Center
End of Sale Date (EoSale)The day the release is no longer orderable or included in manufacturing shipments of Cisco hardware.Approx. 28-30 months after FCS date
End of Software Maintenance (EoSM)This is the last date that Cisco Engineering may release any final software maintenance releases or software fixes for the release.Approx 40-42 months after FCS date

The benefits of Release 15M&T over Releases 12.4 Mainline and 12.4T are:
  • IP Routing
    • Graceful OSPF Restart (RFC 3623) (Helper Mode Only)
    • Graceful Restart for OSPFv3 (RFC 5187) (Helper Mode Only)
    • OSPF Graceful Shutdown
    • OSPF Generic Time to Live (TTL) Security Check (GTSM)
    • Performance Routing (PfR) Protocol Independent Route Control (PIRO)
    • Performance Routing (PfR) EIGRP mGRE DMVPN Hub-and-Spoke Support
    • BGP Graceful Restart per Neighbor
    • Intermediate System-to-Intermediate System (IS-IS) BFD Support
    • IS-IS VRF Support
    • MPLS VPN – Inter-AS Option AB
    • BGP Route Target Changes Without PE-CE Impact
    • IS-IS MIB Support
    • MPLS VPN-BGP Local Convergence
  • IP Multicast
    • IGMP Static Group Range Support
    • IP Multicast Load Splitting – Equal Cost Multipath (ECMP) using S, G and Next-hop
    • IPv4 and IPv6 Multicast Address Group Range Support
    • Multicast MIB VRF Support
    • Multicast VPN Extranet Support
    • Multicast VPN VRF Select
    • PIM Triggered Joins
  • Call Admissions Control Enhancements for Voice and Video
    • RSVP Interface-based Receiver Proxy
    • RSVP Fast Link Repair
    • RSVP VRF lite Admission Control
  • High Availability
    • BFD client for IPv4 Static Routes
    • BFD VRF support
    • BFD Support for WAN Interfaces
  • Embedded Management
    • EEM policy description display
    • EEM policy AAA bypass
    • Multiple CLI execution in one TCL command
  • Security
    • Lightweight IPS Engines for Signatures
    • New Default IOS IPS Category signatures
    • Chaining of Traffic Scanning (Regular Expression) Tables for IPS
    • Configurable Threshold Limits for IPS Signatures
    • GET VPN VRF-Aware GDOI on GM
    • Ability to Disable Volume-based IPSec Lifetime Rekey
    • DMVPN Enhancements
  • Voice
    • Cisco Unified Border Element (CUBE) Support for SRTP-RTP Internetworking
    • CUBE Support for Out-of-dialog SIP OPTIONS Ping Messages to Monitor SIP Servers
    • UC Trusted Firewall Control Version 2
    • Cisco UC Release 8.0 SAF Support

JOb in saudia Arab


Hiring VP IT Operations [Telecom Background, 10+ years exp] Please feel free to contact in order to know more about the opportunities. Contacts:- mklasson@saudinetworkers.com / +966564088570

CCIE training


CCIE Training by CTTC, in karachi

CCIE is the highest level of certification offered by Cisco.
Enhance your skills with the newly revised Cisco CCIE R&S v 4.0
 
Starting Date: Thu Feb 28
Schedule: Mon-Wed-Thu 08:00-09:30 PM
Duration: 3 Months
Course Fee: Rs.14900
Upgrade your CCNP to CCIE for RS 7495
Prerequiste:CCNA Course
Venue: 45 M Block 6, PECHS Karachi
*
The CCIE Routing & Switching Written Exam course download is specifically designed for students who want to focus on the topics and technologies covered in the CCIE Routing & Switching Written Exam version 4 blueprint. This thorough and well-structured course will give you the solid foundation of concepts you'll need to pass the CCIE Routing & Switching Written Exam and also prepare you to continue straight onto CCIE lab exam preparation.
For registrations, please send email to:info@cttc.net.pk
CTTC PECHS Branch
45M, PECHS Block 6, Nursery, Karachi. Tel 34310956-58
Course outline
·         Introduction
·         VLANs,Trunking and VTP
·         Spanning Tree Protocol
·         EtherChannel
·         Frame Relay
·         IPv4 Routing Overview
·         RIP
·         EIGRP
·         OSPF
·         BGP
·         IPv6
·         MPLS
·         Multicast
·         Security
·         Network Services
·         QoS