Cisco Unified CallManager 5.0 presents local SIP enlistment center

Executing SIP progressions in an assembled IP network offers different focal points, including extended worth and customer benefit from new and versatile applications, shipper self-rule for more noticeable determination of employments and endpoints, and the likelihood to diminish costs for gear and the organization of exchanges organizations. Cisco is centered around supporting contraptions that have realized standard SIP-based correspondences to bring to the table customers most outrageous endeavor protection, extended choice through interoperability, and more significant sending decisions call manager. 

Cisco Unified CallManager 5.0 presents nearby SIP selection focus and back to back customer subject matter expert (B2BUA) limits, allowing the affiliation, enlistment, the board, and checking of SIP-based endpoints. The features available to SIP endpoints are practically equivalent to those found in the Cisco SCCP show, allowing SIP and SCCP phones to direct agree and share features. Customers have the chance to move to SIP at their own speed and without the prerequisite for a great deal of customer retraining. Taste is realized locally into focus programming; it isn't executed as an associate course of action, suggesting that the additional cost and unpredictability of discrete applications laborers various vendors use to enlist their SIP phones are not required. Such a nearby consolidation into programming infers it is easier to help and realize, and the costs of keeping a subordinate system are executed. 

The going with SIP-based IP phone models are maintained: 

Cisco Unified IP Phone 7971G-GE 

Cisco Unified IP Phone 7970G 

Cisco Unified IP Phone 7961G-GE 

Cisco Unified IP Phone 7961G 

Cisco Unified IP Phone 7960G 

Cisco Unified IP Phone 7941G-GE 

Cisco Unified IP Phone 7941G 

Cisco Unified IP Phone 7940G 

Cisco Unified IP Phone 7912G 

Cisco Unified IP Phone 7911G 

Cisco Unified IP Phone 7905G 

Any outcast SIP phone, fragile phone, or video endpoint that agrees to RFC 3261 and other related RFCs (should encounter the SIP Verified program if affirmation of interoperability is needed); more information about the SIP Verified program is arranged at: http://forums.cisco.com/eforum/servlet/IPCApps;jsessionid=zhyxvhy9f1.SJ2B?page=tpsipep 

Computer correspondence compromise (CTI) maintain for SIP phones—Cisco Unified CallManager 5.0 engages CTI applications to control and screen SIP phones along these lines they have as of late used for SCCP phones. 

SIP trunk-side enhancements—Version 5.0 familiarizes immense redesigns with SIP trunk-side hailing help and call guiding capacities through assistance for the latest RFCs and Internet drafts, including yet not confined to: SIP (RFC 3261), the REFER method (RFC 3515), the REPLACES header (RFC 3891), the Remote Party ID (RPID) header, the Unsolicited NOTIFY procedure, and the SUBSCRIBE/NOTIFY strategy. Taste trunk-side frameworks organization gives an open interface to applications to interface to the Cisco Unified CallManager 5.0. 

REFER maintain—Supported by Cisco Unified CallManager 5.0 for SIP-began moves; these trades are done from the Cisco Unified CallManager or from an outcast application 

Replaces header—This improvement is used to displace a current SIP trade with another; the ability to change the header is a component of the Cisco Unified CallManager being a B2BUA. 

Subscribe/illuminate event declaring—This redesign is open on the SIP trunk for use by external applications, for instance, the Cisco Unified Presence Server to report presence to an external component.

voice engineer work

Set of working obligations 

The Sr. Voice Engineer will responsible for driving method and movement of complex Voice and Trader Voice projects. Reliable to lead work effort for nonstop essential exercises in districts of Voice recording and Unified Communications. This individual will be working personally with other specific topic specialists and organizers, similarly as security and consistence, plan and creation the board specialists obligated for transport of an ensured, pleasant and stable endeavor UC stage. Supervising complex endeavors inside association organization underwriting and ensuring productive use. 

Commitments/Duties: 

This individual will be responsible for execution voice engineer job and courses of action movement of Unified Communications things and updates reliant on current creation associations for Voice over IP and Trader Voice system 

Give VoIP and Trader Voice organizations maintain at a mid to forefront level to the endeavor 

Perform mid to bleeding edge level Cisco UC association, execution and examining of CUCM, CUC, Meeting Place, UCCX, CUPS, CUBE, CER, CVP, Jabber, Gateways MGCP, H323, SIP, CTI blend to trading stages, dialers, call recorders and work territory stages, and supporting structure 

Give Planning, Design and Implementation all through the present Lifecycle, including application and hardware redesigns, and course of action smoothing out 

Study and plan system arrangements and voice necessities for NICE Recording structure for Call Manager and IPC associations

Work personally with the picked Voice Recording vendor, (NICE/Cybertech) to study structure conclusions and the connected arrangements 

Surrender and keep compositional documentation of transmission correspondences system and making handover documentation to Service Operations 

Research organization/VOIP issues, insinuating shippers or experts as appropriate 

Liable for realizing and keeping up all pieces of the VOIP network structure (for instance Cabling-Racking/LAN/WAN/WLAN/VoIP/VPN/Network Security) 

Participate in multi-discipline design gatherings to give plans and extension association of equipment and VOIP association 

Give Tier 3 Team quickening and interface w/Vendors to decide all issues changing 

Execute vital plans to meet tight schedules and to ensure irrelevant unsettling influence to the Customer environment giving World Class Customer Service to inside and outside customers 

Give VoIP and association scene and issue the heads 

Help with progression of documentation for standards necessities inside the workplace 

Give exact, clear, brief documentation and help to Service Operations 

Review and assurance exact structures documentation is kept up over the range of the errand 

Phenomenal agreeable individual with ability to encourage and assist others and give junior Team part mentoring 

CUCM coordination with Microsoft Lync, and IPC systems 

Host project meeting with key staff to review project progress, disseminate status gives a record of a booked reason 

Capacities: 

Critical level data and association in Call Manager 8.6 or higher, ace level capacity in utilizing Bulk Administration Templates and Tools. 

VMWare vSphere Experience preferred 

Association in organizing and researching Cisco VoIP structures, programming, and gear including Call Manager, Unity and IPCC (Express, etc… 

Dynamic association with Call control, IP directing, VOIP game plans and SIP Trunking are principal 

Inclusion in IPC turrets, Speaker transport, Nice Voice Recorders is at least an 

Strong Cisco coordinating and trading/data/voice sorting out capacities including H323/SIP/MGCP/MTP, etc… 

Least of 3-5 years dynamic association in Cisco Call Manager/Unified Communications Manager and Cisco Unity 

Current Cisco Voice affirmations for instance CCVP solidly loved 

Vendor Voice IPC and SpeakerBus experience is needed including turrets and expansions

cisco voice engineer occupations

Set of working obligations: 

Overall Information Grid (GIG) Service Management-Operations (GSM-O), a Defense Group of Leidos has an opening for a Cisco Voice Engineer. Our customer is the Defense Information Systems Agency (DISA) arranged at Scott AFB, IL OR Hill AFB, Utah and goes probably as the provider of GIG/Defense Information System Network (DISN) organizations to the division of Defense (DoD) and public security affiliations. 

Fundamental Responsibilities: 

Sponsorship keeping an immense, complex and especially solid voice establishment. 

Contribution in Cisco Voice stages, Cisco Unified Communications Manager (CUCM), Unity and Cisco Unified Boundary Element (CUBE). 

Administer, mastermind, examine and ensure cisco voice engineer jobs organization supporting all pieces of a Cisco voice and data association. Ought to be facilitated, report orchestrated, have capacities in Visio sketching out, and understand WAN chiefs. 

Additional Duties and Responsibilities: 

Make particular course of action and work bearings and reporting on an endeavor IP correspondence association. 

Research and resolve complex particular issues utilizing ace data on SIP hailing and call-streams. 

Explore far away site enlistment, hailing and media issues. 

Organize course plans, course records, contraption profiles, calling search spaces, and course portions. Affirm change requests and assurance precision. 

Supervise research and organize Cisco Unity Connection features, allies, call regulators and call coordinating rules. 

Manage examine and organize Cisco or possibly Juniper coordinating and trading rules. 

Essentials: 

Requires BS degree and 2-4 years of prior appropriate insight. Additional experience, tutoring and getting ready may be considered in lieu of degree. 

Capacity in SIP internetworking 

Plainly obvious experience of bleeding edge examining to detach and dissect network issues for directing, trading, security and gigantic extension voice executions. 

DoD 8570 IAT II (Security+) 

At present have a working Secret outstanding status. 

Needed capacities: 

Working data on Secure Real-Time Transport Protocol (SRTP) including related G.711 and G.729 codecs and Digital Tone Multi Frequency (DTMF). 

Contribution in Oracle SBCs and other voice media gateway machines. 

Experience supporting passed on firewall structure. 

Experience making/supporting IPSec MPLS VPN system *Experience completing and supporting BGP directing shows *Experience arranging, planning and examining QoS gathering, checking and prioritization of association traffic (voice, fundamental applications, etc) 

Information on IPSec execution and researching. 

Information on Unimax natural motorized provisioning, work cycles and survey uncovering.

Configure CUCM for Interoperation with Skype for Business Server

The settings described here are meant only as examples of how CUCM can be configured to work with a VIS. Other settings and/or usages of alternate CUCM functionality could also be used to achieve the same result. No recommendation is implied as to the optimal configuration for a particular scenario call manager.

A number of CUCM settings need to be confirmed or changed for interoperation with the VIS. Follow the procedures below in order to avoid missing required settings.

Configure the CUCM

Log in to CUCM and navigate to Cisco Unified CM Administration->Call Routing->Class of Control->Partition.

In the Partition Configuration screen, enter the partition name and description and click on Add New.

Navigate to Cisco Unified CM Administration->Call Routing->Class of Control->Calling Search Space.

In the Calling Search Space Configuration screen, enter the name for the calling search space, and in Selected Partitions, enter the name of the partition you just created. Click Save when done.

Navigate to Cisco Unified CM Administration->System->Security->SIP Trunk Security Profile.

In the SIP Trunk Security Profile Configuration screen, set the SIP Trunk Security Profile Information options as shown, and click on Add New.

TABLE 1

Parameter Recommended setting

Name

SfBVideoInterop_SecurityProfile

Device Security Mode

Non Secure

Incoming Transport Type

TCP + UDP

Outgoing Transport Type

TCP

Incoming Port

5060

Navigate to Cisco Unified CM Administration->Device->Device Settings->SIP Profile.

In the SIP Profile Configuration screen, set the SIP Profile Information options as shown.

TABLE 2

Parameter Recommended setting

Name

SfBVideoInterop_SIPProfile

Description

SfBVideoInterop_SIPProfile

On the same screen, scroll down to the SDP Profile Information section. The SDP Session-level Bandwidth Modifier for Early Offer and Re-invites option is set by default to TIAS and AS. Change this option to TIAS only. If you leave this option at its default setting, Skype for Business Server will not understand the bandwidth modifier information in the SIP message. TIAS means Transport Independent Application Specific while AS means Application Specific. These are SIP options specified in RFC3890.

On the same screen, scroll down further. Under the SIP Profile's Trunk Specific Configuration, select Early Offer Support for voice and video calls and set it to the Mandatory (insert MTP if needed) option. This will enable CUCM to set up an outgoing SIP call with Early Offer. One new feature in CUCM 8.5 and beyond is that it supports outgoing call setup with Early Offer without requiring Media Termination Point (MTP).

Verify that in the SIP Options ping section, the box is checked next to "Enable OPTIONS Ping to monitor destination status for Trunks with Service Type 'None (Default)'."

When you are finished, click on Add New.

Navigate to Cisco Unified CM Administration->Device->Trunk.

Set the Device Protocol to SIP and press Next.

Under Device Information, Set the Device Name and Description (probably to something like SfBVideoInterop_SIPTrunk), and set the Media Resource Group List to an MRGL that contains the right media resources.

Scroll down further. Media Termination Point (MTP) is not required for Video Calls, if it is not already unchecked, uncheck it. Check the option to Run on all active Unified CM Nodes. Please note that you should add all CUCM nodes to the Skype for Business Server configuration.

Scroll down further. Set the Inbound Calls and Connected Party Settings options as shown.

TABLE 3

Parameter Recommended setting

Calling Search Space

CSS_SfBVideoInterop

AAR Calling Search Space

CSS_SfBVideoInterop

Connected Party Transformation CSS

CSS_SfBVideoInterop

Scroll down further. Under the SIP Information Destination section of the SIP Trunk configuration, specify the VIS Pool's FQDN or the IP address of individual VIS servers in the pool (adding multiple entries). In the Destination Port specify the Port that VIS is listening at for connections from CUCM (the default is 6001). Also specify the SIP Trunk security profile and SIP profile you created earlier, as shown.

TABLE 4

Parameter Recommended setting

SIP Trunk Security Profile

SfBVideoInterop_SecurityProfile

Rerouting Calling Search Space

CSS_SfBVideoInterop

Out-of-Dialog Refer Calling Search Space

CSS_SfBVideoInterop

Subscribe Calling Search Space

CSS_SfBVideoInterop

SIP Profile

SfBVideoInterop_SIPProfile

DTMF Signaling Method

RFC 2833

Scroll down further. Set the Recording Information as appropriate for your system. It's fine to leave it set to None.

When you are finished, click on Add New.

Navigate to Cisco Unified CM Administration->Call Routing->Route/Hunt->Route pattern.

In the Route Pattern Configuration screen, enter the Pattern definition parameters shown below. Scroll down to the Called Party Transformations section and set the mask as shown, and then click on Add New when finished. 

Fog computing: what role for the IoT?

Definition of fog computing

Fog computing, also called "computing in the fog", defines an infrastructure responsible for storing and processing data from connected objects . Direct competitor, alternative or complementary solution to cloud computing, fog computing has the particularity of storing and processing data through the use of equipment located at the edge of the network. It therefore makes it possible to carry out these two actions locally, without having to request a datacenter located several hundred kilometers away or a cloud. In this area of ​​storage and processing of data and IoT, fog computing creates an additional interface that can be situated between Edge Computing and Cloud Computing.

Fog VS edge computing

Fog computing and edge computing are two quite similar infrastructures. Both are based on the processing of data produced directly by objects connected at the edge of the network. This proximity to the point of origin leads to a significant reduction in latency (more travel between the connected object and the cloud). The difference between fog and edge computing is the IT equipment involved: according to the OpenFog Consortium, edge refers to processing terminals while fog computing refers to IT architecture. In addition, we speak of edge computing when the computing resources are in the connected object, and fog computing when they are in a separate network node, such as an IoT gateway. 

Fog computing actively participates in decongesting network traffic (only the most important information is transmitted to the company's servers) and in increasing its performance (notably by reducing bandwidth consumption). Finally, it should be noted that fog computing is also intended to strengthen data security. By storing them securely locally, it limits possible intrusion attempts and prevents cyber attacks against the cloud or data centers.

Cisco voice engineer jobs

It is now accepted to say that we owe the concept of fog computing to an engineer at Cisco Systems, an American computer company specializing in servers. Since 2015, there has also been the OpenFog consortium (which brings together companies such as Cisco, Dell, Microsoft and even ARM) responsible for defining network protocols and technological standards useful for market standardization. The objective: to succeed in giving a definition of distributed architecture that can be able to adapt to the very strong development of the IoT. On the technical level, the fog computing infrastructure is made up of several layers, respecting the stacking logic that is already found in the cloud, into which its own elements are then integrated,

Call management via the Cisco Business Edition 6000 solution

Call handling by the Cisco BE 6000 solution

The solution allows all types of call processing necessary within a company: call manager

Advanced telephone functions (unified messaging, chat and presence, contact center, extended video, web conferencing, attendant console, high availability)

All the features of ToIP

Voicemail and pre-hook

Integrated productivity apps

Complete portfolio of Cisco IP phones to meet all SMB needs

Unique and common management solution

Solution including switching, VPN, security, data routing

Call management by the Cisco BE 6000 solution

Advanced IP convergence features

Video Telephony on the workstation

SIP trunk

Integration of WiFi 802.11n

SoftPhone

Tele worker

Integrated voicemail

XML Services Support on Cisco IP Phones

Interactive voice server

Unique number

Click to Call

Integration with the company directory

Wireless mobility with wifi and / or dual mode terminals

And all the traditional telephony features:

Transfer on busy, no answer and mandatory

Do not disturb

Support double call per line

Hold, park and transfer a call

Call grouping, call pickup

Caller name and number display

Intercom, Supervision

Music on hold, Night service

Conference

Internal / External differentiation of the ringer

Missed, made and received calls log

Recording conversations

The modes of access to his voice mail

Unified communication at the heart of all exchange processes is undoubtedly a time saver in productivity and working comfort. You can therefore consult your voicemail using the following different access modes:

From his landline

Visual Voice Mail makes it possible to navigate graphically in the interface to consult the messages left at any time.

From the unified client

The Cisco Jabber or Lotus Sametime client or even Lync are just some of the ways to access your voice messages.

From the Email client

The messaging clients can be used to consult its voicemail messages but also to classify and archive them. The transfer is done by the IMAP client or by RSS feed. In this case, the Business Edition 6000 solution ensures consistency between text and voice messages, an undeniable advantage in many cases and a substantial gain in productivity for employees.

In mobility

Mobility situations are not to be outdone. It is of course possible to consult your voicemail messages from your mobile via the Personal Communications Assistant (PCA) web portal, or the unified messaging web portal or the Jabber Client.

Critical bug 9.9 / 10 at Cisco: Quickly fix this hole in Jabber for Windows and MacOS

Watchcom reported four vulnerabilities to Cisco earlier this year

According to Cisco, the bugs allow an attacker "to run arbitrary programs on the underlying operating system with elevated privileges or to access sensitive information." Customers have no choice but to install the latest updates to prevent attacks voice engineer jobs.

Norwegian security firm Watchcom discovered earlier this year that Jabber was vulnerable to a cross-site scripting (XSS) attack through XHTML-IM messages. Jabber did not properly sanitize incoming HTML messages and instead passed them through a faulty XSS filter.

Cisco notes that new message processing vulnerabilities can be exploited if an attacker can send Extensible Messaging and Presence Protocol (XMPP) messages to end user systems that are running Cisco Jabber. "Attackers may need access to the same XMPP domain or another access method to be able to send messages to clients," Cisco notes in a notice .

The three partially fixed bugs are tracked as CVE-2020-26085, CVE-2020-27127, and CVE-2020-27132.

Watchcom reported four vulnerabilities to Cisco earlier this year, and they were revealed by the network giant in September. But three of them weren't properly fixed in updates at the time, according to Watchcom.

Failed mitigation

Watchcom reviewed the fixes after a customer requested an audit to verify that the bugs had been sufficiently mitigated in existing Cisco fixes. He found that the bugs had not been mitigated.

Two of the three poorly fixed bugs can be used to achieve remote code execution. One of them can also be used to get NT LAN Manager (NTLM) password hashes from users.

“Two of the vulnerabilities are due to the ability to inject custom HTML tags into XMPP messages,” says Watchcom penetration tester Fredrik Bugge Lyche. “The patch released in September only patched specific injection points that Watchcom had identified. The underlying problem has not been addressed. We were therefore able to find new injection points that could be used to exploit vulnerabilities