Another $1400 lunch coupon, please...

Monday, July 8, 2013

3 Months After my CCIE

Well its been 3 months after passing my CCIE and life has changed a little. The most dramatic change has to be the expectation to know the exact answer to everything voice all the time. Its pretty crazy but I wouldn't trade it for anything. It was a very long and hard road and I want to thank Cisco for making it right after the Collaboration conversion. I think now I want to transition my blog to be more of a contact center blog.

* (Side Note) If you think you know voice or want to actually push the boundaries of your knowledge. You need to transition into contact center. Its is continuously challenging.

Tuesday, March 20, 2012

IpExperts WB2 LB1

First 8 hour lab back after a long hiatus, and it was not too bad. I got a 74 out of the attempted 87 points. Initial problems are listed below and there was a lot of room for improvement. I did how ever take a different approach to attacking the lab. Instead of break each section down and trying to attack them with set times for completion, I looked over the exam for any glaring things that might cause problems and then walked directly through the lab. I finished up earlier than anticipated and had time to trouble shoot issues that I skipped over. (I need to work in on walking away from a problem.)

· Lan /Wan Qos - I spent time doing this but always end up missing one thing that cost me the points.

· Attendant Console – Finally got it working at the end but forgot to put a Final Route for directing to voicemail.

· UCCX – Integration problems – (was unable to view axl on ucm with client that had axl privileges. )

· Voice-Mail Service – Need to be able to rebuild voicemail service if it has been deleted. IE UCM 7.X Release Notes.

· IPIPGW – Was not a problem but took more time than needed too. This was primarily because I slowly walked the call through instead of building XCoders and DialPeers in the IPIPGW before making call.

So Final on Lab 1 was a 74

Thursday, March 15, 2012

CME – Sip Endpoints / SKINNY Endpoint and Router and the headache of Synchronizing Clocks

First let my start off by saying this is something that if you have never run into doesn’t seem to be a daunting task and once you understand all the different components it isn’t. But getting from point A to B is not always ice cream and xbox. To properly trouble shoot the problem me must first understand the basics of where the time is pulled from each one of the devices.

Router – Local and or NTP server

Sip Endpoint – NTP server

SKINNY Endpoint – CME

Remembering these simple things and you can alleviate a lot of stress later on when one set of devices are off by an hour. And when you start to take into account daylight savings time (DST) requirements it throws another clown into the party. Here are the individual configurations needed for each of the device types for congruent clocks.

Router

- Local – #ntp master

- NTP server - #ntp server X.X.X.X

- Clock time zone - #clock timezone GMT +23/-23

- DST – clock summer-time DST recurring

Skinny Endpoints in Telephony-services

- Time –zone - time-zone ( * make sure you are 100% on the DST of the location you are specifying as this can cause headaches)

SIP Endpoints in Voice Register Global

- Time-zone – timezone (*make sure you are 100% on the DST of the location you are specifying as this can cause headaches )

- Ntp-server – ntp-server X.X.X.X (Usually the same NTP server that the router is using. I have had problems with using the NTP master and sourcing the local CME box.)

Simple Right? Well it is now, good luck and have fun studying.

Saturday, February 18, 2012

T.30 Faxing Phases

Phase A—Establishing a Voice Call

The call originator prepares a fax and dials a destination number. The destination fax device picks up the call. The originator and the destination are now connected in a voice call, but to transition to fax transmission one party must signal that it is a fax device. Either device can send its signal first, using one of the following methods:

• The calling device sends a Calling Tone (CNG) to the destination device. The CNG identifies the calling device as a fax machine. The CNG is a repeating 1100-Hz tone that is on for 0.5 seconds and then off for 3 seconds.

• The called device sends a Called Station Identifier (CED) tone, which identifies the called device as a fax machine. CED is a 2100-Hz tone that is on for 2.6 to 4 seconds.

Once these messages have been exchanged, the transaction can move to phase B.

Phase B—Identifying Facilities and Capabilities

The following sequence of events identifies facilities and capabilities for fax transmission:

1. The called device sends a Digital Information Signal (DIS), which describes the called fax machine's reception facilities, such as maximum page length, scan line time, image resolution, and error correction mode. Many standard facilities are contained in the DIS message, and they are defined in the T.30 specification.

2. The calling device examines the DIS message and in response sends a Digital Command Signal (DCS) that tells the called device which facilities to select for the reception of the fax transmission.

3. The called device may also choose to send the following optional messages:

• Called Subscriber Identification (CSI) provides some detail as to the identity of the called device.

• Non-Standard Facilities (NSF) informs the calling device that the called device may have some extra features that can be utilized during the fax transmission.

4. The calling device can then choose to send a Transmitting Subscriber Identification (TSI) message. Also, in response to an NSF message, the calling device can send a Non-Standard facilities Setup (NSS) message to select extra reception parameters on the called device.

5. The calling device now sends the Training Check (TCF) message, which includes a stream of 0s for about 1.5 seconds through the HS modulation that was agreed upon during the DIS-DCS handshake. The called device then responds with a Failure To Train (FTT) if the modulation speed is not acceptable or with a Confirmation to Receive (CFR) if the modulation speed is acceptable. Training is a process that verifies the communication path.

6. Once the training has been completed and the modulation speed is agreed upon, the fax devices move to phase C and start the transmission of T.4 page data using HS modulation.

Phase C—Transmitting Content

Phase C is referred to as the In-message Procedure. During this phase, high-speed T.4 page data is sent one line at a time. Each burst of line data is followed by an End Of Line (EOL) message. Because the EOL information is sent as T.4 data, it would not necessarily be seen in a T.30 trace. When the sending device has finished sending pages or wishes to return back to control mode, it sends 6 EOLs in a series that constitutes a Return To Control (RTC) message. The RTC message indicates the end of phase C, and the call progresses to phase D.

Note If the fax machines decide during phase B to use Error Correction Mode (ECM), the format of the data sent during phase C may be different. With ECM, the T.4 page data is grouped into high-level data link control (HDLC) frames rather than being sent in a raw stream. This means that if the HDLC blocks of T.4 page data are not received error-free, a Partial Page Request (PPR) message can be sent, listing the frames that were not received and asking for them to be resent.

Phase D—Signaling End of Transmission and Confirmation

After the T.4 transmission and the subsequent return to control mode, the sending device must send one of the following signals:

• Partial Page Signal (PPS)—Devices that send faxes with ECM can send a PPS, which must be acknowledged by a Message Confirmation (MCF) signal from the receiving device.

• End Of Procedure (EOP)—This signal indicates that transmission of pages is complete and that there are no more pages to send. The EOP must be acknowledged with an MCF from the receiving device, after which the devices can move to phase E.

Phase E—Releasing the Call

Following the fax transmission and the postmessage transactions, either the calling device or the called device can send a Disconnect (DCN) message, at which point the devices tear down the call, and the telephony call control layer releases the circuit. DCN messages do not require a response from the opposite device.

This was pulled from Cisco Fax Service over IP Applications.

Wednesday, August 24, 2011

Major 15.1(3)T Voice Features

RSVP Ingress CAC (Connection Admission Control)

• With load balancing, inbound traffic for the Resource Reservation Protocol (RSVP) agent is not guaranteed to use same WAN link as outbound traffic. This traffic can overrun the priority queue because reservations are currently made only on outgoing CE (Customer Edge) interfaces, in the direction of the Real-Time Protocol (RTP) stream. The Ingress CAC mechanism improves the solution by performing CAC on the ingress interface on the next hop.

Benefits

• RSVP can now work with asymmetric media paths (asymmetric routing cases).

• RSVP can now meet the needs of headquarters-to-branch-office deployment scenarios in which bandwidth profiles are not identical.

• RSVP is now aligned with service provider deployments.

Mediatrace 1.0

• Mediatrace is a powerful tracing and dynamic configuration tool. It provides automatic path discovery of traffic streams and can discover both Layer 2 and 3 devices along the path. Mediatrace while following the path of particular flow can gather crucial hop by hop information like packet loss and network jitter that the flow might be experiencing. It can also be used to collect system information such as CPU and memory utilization data of the intermediate hops. Mediatrace can be run on demand or scheduled for periodic monitoring and data collection.

Benefits

• Mediatrace provides powerful troubleshooting and fault isolation from a single point in the network.

• Dynamic configuration provides an efficient monitoring mechanism to help reduce system resource consumption for network devices.

• Support for both Layer 2 and 3 devices provides visibility on the complete path.

Cisco Unified Communications Manager Express (UCME) and Selected Survivable Remote Site Telephony (SRST) 8.5 Enhancements

• The main Cisco UCME and SRST features delivered in Release 8.5 release of the products include support for the following:

• Overlap sending on ISDN and Primary Rate Interface (PRI)

• Forced authorization code (FAC)

• Signal-to-noise ratio (SNR) enhancements on Cisco Unified Communications Manager Express

• SSL VPN client on SCCP UCME Customization of services and directories page

• Immediate divert (iDivert) feature on Session Initiation Protocol (SIP) phones

• Busy lamp field on phone applications

Benefits

• Release 8.5 of Cisco Unified Communications Manager Express and Cisco Unified SRST reduces the cost of Cisco Unified Communications Manager Express teleworkers through improved deployment flexibility of SSL VPN clients on SCCP Cisco Unified IP Phones.

• This release also improves the Cisco IP Phone user interface and the billing and accounting options, and it makes Cisco Unified Communications Manager Express and Cisco Unified SRST easier to deploy and maintain.

Cisco Advanced Foreign Exchange Station (FXS) Analog Gateway Features

• This release adds advanced FXS Analog Gateway features and SCCP over Transport Layer Security (TLS) with Cisco Unified Communications Manager Express.

• The security endpoint uses the existing PKI, which is used to connect to Cisco Unified Communications Manager Express. The PKI obtains the certificate from a third party certificate authority (CA) and uses it to establish TLS sessions with Cisco Unified Communications Manager. Secure RTP (SRTP) is used to encrypt the media.

• Advanced supplementary features on FXS analog ports with Cisco Unified Communications Manager include:

o Repetitive call-waiting tone

o Callback on no answer

o Media renegotiation

o cBarge

o Configurable exclusive audio message waiting indicator (AMWI) and visual message waiting indicator (VMWI)

o Single-number reach (SNR) (supported for both Cisco Unified Communications Manager and Cisco Unified Communications Manager Express)

• Enhanced serviceability for analog FXS voice port line measurement on Cisco VG224 Analog Voice Gateways and Cisco IAD2430 Integrated Access Devices and certain voice interface cards will be added. New interfaces will be created to measure line voltage, current, impedance, and capacitance of the analog FXS voice port.

Benefits

• The capability to provide granular troubleshooting capabilities during initial network provisioning improves deployment when disparate analog endpoints are included.

• Automated signaling and media continuity checks make network operations and health monitoring easier in large analog deployments.

Cisco ISR SIP Gateway and Cisco Unified Border Element 8.6 Enhancements: Reporting End-of-Call Statistics in SIP BYE Message

• Cisco Unified Border Element and the Cisco ISR SIP voice gateway report end-of-call statistics through the SIP BYE message.

Benefits

• This gateway can provide demarcation functions on the public switched telephone network (PSTN) gateway as well as the Cisco Unified Border Element. These statistics can be used to update call data records on Cisco Unified Communications Manager or Cisco Unified Communications Manager Express

Cisco ISR SIP Gateway and Cisco Unified Border Element 8.6 Enhancements: Conditional SIP Header Manipulation

• Cisco Unified Border Element 8.6 adds support for manipulation of one SIP header based on the contents of another SIP header. This feature extends the SIP profile function that is already available on the Cisco Unified Border Element. It includes the capability to copy the contents of one header to another, or modify the contents of one header based on the contents of another. SIP headers on the out-leg can be modified based on the contents of any SIP header on any SIP message received on the in-leg.

Benefits