Posts

How to make Collaboration Audio Dial Plan

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Folks, In this post, I am sharing the dial plan topology that I use to make for all my accounts. This particular sample shows dial plan based on migration strategy. Real Quick on the setup - We have multiple Call Manager Express Sites , Call Manager Clusters as well as Session Manager Cluster currently. Proposed design - consolidate all sites into one Unified Communications Manager Cluster. This dial plan talks about how to achieve the proposed design leveraging phased migration approach. That would mean as we migrate the sites , inter site dialing between the migrated sites and old sites would leverage SME ILS/GDPR mechanism to advertise dial plan and route the call. Take a look below- This is the sample dial plan for your reference, enjoy!

Music On Hold - Decision, Allocation, Troubleshooting

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Decision Tree for IP Voice Media Streaming Service on what resource to allocate What Audio File and MRGL to choose  Music On Hold SIP Trace - Troubleshooting reference

First non-technical post - passport lost

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

Sample configuration for Video Conference Call through CUBE

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

P2P video call

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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.

SIP Phone Registeration on CME

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

Important Service Parameters in CUCM

" 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.