Debugging VOIP with Shikha Sharma #CiscoKid

This is the first time i am writing a non-technical post in a technical blog only because I don't have any other blog to post this information. I was traveling USA on a B1 and have lost my passport in a theft. I could not find anywhere what next should I do. Hence, writing for those of you who might get into this problem. I found some embassy numbers to call but none of it answered / helped me with right information. Washington embassy - 2029397000 Atlanta Embassy - 404 963-5902 NY Consulate - 8459990726 Overall, I was being told that I would need to go to the embassy which belongs to my region and submit documentation there. The information of your region embassy and documentation are mentioned in this link- http://www.blsindia-usa.com/passport/lost_passport.php Mainly, you'd need NRI document to be filled (even though you are not) alongwith your passport photo. There are two options you can opt for - 1. Emergency travel document - using this document you can

Here is the Sample configuration for Video Conference Call through CUBE voice-card 0  voice-service dsp-reservation 50 ====> dsp needs to be reserved for VOIP calls and TDM video calls, %age of dsps. Unreserved DSPs can be used for IP video. If you reserve no DSPs for VOIP calls, you'll not be able to add video as it needs VOIP audio. There will be no way to even add the codecs. ! dspfarm profile 2 conference video homogeneous  codec ilbc  codec g711ulaw  codec g711alaw  codec g729ar8  codec g729abr8  codec g729r8  codec g729br8 ====> all the audio codec will be added by default  codec h264 cif frame-rate 30 bitrate 320kbps ===> all participants will be forced to use same codec, same framerate and bitrate, if negotiation fails, call fallback to audio only. Hetergenous mode- participants can use their own codec/bit/frame rate and multiple video codecs can be added however, it requires more PVDM3 resources as it reserves the dsps for video conference.  maximu

In order to enable video capabilities on the phone, you should have "video capabilities" field as enabled on the phone's configuration page. There is no other configuration required on CUCM and gateways.

voice service voip sip ===> enables SIP on the router registrar server ===> Accepts registration requests ! voice register global  mode cme  source-address <ip address of cme router> port 5060 ===> IP address through which Phones will communicate with CME  max-dn 10 ===> count of DNs that can be added  max-pool 1 ===> count of pool of either a single IP phone or group of phones. ! voice register dn  1 ===> where you create the number to assign on Phones  number 4002 ! voice register pool  1  id mac 580A.2099.3F37 ===> using the pool to create a single IP Phone pool  type 9971 ===> defining type is as important as defining the mac-address  number 1 dn 1 ! voice register global create profile ===> similar to create cnf-files, this will create the phone profile. ! end

" always use transformation for remote party" Cisco IP Phones do not understand the transformation patterns even though you have applied its CSS in their device pool. This parameter should be enabled so that Phones start to take the applied transformation patterns. This parameter is efficient in a global dial plan to normalize the calling number displayed on the IP Phone. "always display the original dialed number" As the name suggests, this parameter is used to display the original dialed number on the IP Phones, even though the appending digits or dialing code gets stripped.

RTR1#show inven NAME: "CISCO2951/K9", DESCR: "CISCO2951/K9 chassis, Hw Serial#: FGL170512NA, Hw Revision: 1.1" PID: CISCO2951/K9      , VID: V05 , SN: FGL170512NA NAME: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 SubSlot 0", DESCR: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1" PID: VWIC2-2MFT-T1/E1  , VID: V01 , SN: FOC16520EUD Linecode, framing and switchtype information need to match with Service Provider. Please have this information ready before configuring the PRI. 1. Bring up the T1 Voice card RTR1(config)#card type t1 0 0 2. A llow the ports on this WAN interface card (VWIC) to use the network clock for timing RTR1(config)#network-clock-participate wic 0  3. Define the ISDN switchtype globally. RTR1(config)#isdn switch-type primary-ni 4. Configure controller controller t1 0/0/0 pri-group timeslots 1-24 linecode b8zs framing esf

To simulate test outbound call on the router- R1#show dialplan number <test number> This command will show you which dialpeer will be hit if this number is dialed. Example- RTR1#show dialplan number 911 Macro Exp.: 911 VoiceOverIpPeer15400 peer type = voice, system default peer = FALSE, information type = voice, description = `CME1-TO-CME2-911', tag = 15400, destination-pattern = `911$', voice reg type = 0, corresponding tag = 0, allow watch = FALSE answer-address = `', preference=1, To test translation rules- R1#test voice translation-rule <rule#> <number to translate> Example voice translation-rule 1  rule 4 /^914102850417/ /DEAD/  rule 5 /^91410915..../ /DEAD/  rule 36 /9011204/ /000204/  rule 37 /9011226/ /000226/ RTR1#test voice translation-rule 1 914102850417   Matched with rule 4 Original number: 914102850417 Translated number: DEAD Original number type: none Tr

Basic configuration of DHCP on a router- By deafult, DHCP server and relay agent service is enabled on a router, hence, we do not need to start it. 1. Instead of mentioning the range which should be assigned as IP address for the devices using your DHCP server, we mention the address it should exclude to assign to the devices- R1(conf)#ip dhcp excluded-address <range of IP address> example R1(conf)#ip dhcp excluded-address 142.100.10.20 142.100.10.50 R1(conf)#ip dhcp excluded-address 142.100.10.80 142.100.10.254 This means only range 142.100.10.1-10 and 142.100.10.51-79 is available for the devices. 2. Configure the DHCP Pool R1(conf)#ip dhcp pool POOL1  ---- define the name of the pool R1( dhcp-config)# network 142.100.10.0 255.255.255.0 -- define the IP&mask of the DHCP pool's network R1( dhcp-config)#default router 142.100.10.20  -- define default gateway R1( dhcp-config)#option 150 142.100.10.20 -- define TFTP address 3. Configure IP helper Addres

NTP configuration on the router is pretty simple. Make your router as NTP server - 1. Set the clock on the router. You can use "clock" command in the privilege mode to set the time. 2. In order to configure it as master, configure the router as - "ntp master <stratum>". The stratum n umber from 1 to 15. Indicates the NTP stratum number that the system will claim. This will make your router as master. 06:49:10.078 UTC Sun Mar 22 2015 ntp master 5 NTP client configuration- 1. Configure ntp master server address in the privilege mode- ntp server <ip address of ntp master> Validating NTP associations - Check its clock R1#show clock 06:58:15.888 UTC Sun Mar 22 2015 R1#show ntp associations   address         ref clock       st   when   poll reach  delay  offset   disp *~157.26.1.254    .LOCL.           1   1021   1024   377  0.706  -0.926  0.054  * sys.peer, # selected, + candidate, - outlyer, x falseticker, ~ configured

Web Dialer is the feature which helps you to dial through web using your IP Phone. Its configuration is pretty simple as the application user is auto created. All you need to do is- 1. Enable the web dialer service. 2. Go to Service Parameters > web dialer - Provide CTI Manager Primary and Secondary Servers. This would require restart of the service. 3. Assign "Standard CTI Enabled" user group on your End User Configuration. Your end user should have your IP Phone as controlled device. 4. Access web dialer service : https://<IP-Address-CTI-Manager>/webdialer/WebDialer 5. Enter the username and password and you are done. Enjoy the web dialer service!