Index

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Dashboard.................................................................................4

Agents........................................................................................5

SIP...........................................................................................14

Queues.....................................................................................18

Trunk........................................................................................24

Dial Plan..................................................................................29

Audio.......................................................................................35

Cally Square.............................................................................36

Pre Analytics............................................................................37

Settings....................................................................................39

Zendesk Integration.................................................................42

IntroductionxCALLY Shuttle is the next xCALLY generation software suite, providing many key benefits if you

are looking for a professional customer care solution for Asterisk.

Here are few of them:

- Responsive Supervisor web interface HTML5

- Support for 3 type of agent experiences:

- Windows CTI phone bar

- External CTI SIP phones

- WebRTC (experimental)

- Cally Square Drag and Drop IVR full Web HTML5 Asterisk IDE

- Integration with 3rd part software (i.e. Zendesk) using the Shuttle Push Technology

- Advanced reportings

- Advance call routing management

- Calendar

- Linux CentOS 6.X super-easy installer.

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Dashboard

In the Dashboard section there is an overview on the state of the system, in particular:

- real-time monitoring of the events that may occur, such as the expiry of a timeline

- list of the open activities, with their progress

- list of messages of the integrated messaging system.

On the right of the screen, on the top bar, there are two buttons useful to:

- change the language

- logout from xCally Shuttle .

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AgentsxCALLY Shuttle Agents are called operators. In this last generation system there is an important distinction between the users of the system. Agent: The so-called call center operator, associated with a SIP only. User: The system administrator for the configuration and management. SIP: The actual SIP account (which can also be Web).

In the Agents section you can manage the call center agents.

In this screen there is the list of the agents already created, with their main informations.

There are three actions that can be done quickly through the Agents table: - Send an e-mail to the Agent by clicking on the button - Delete the Agent with a click on the button - Change the parameters specified in the creation of Agent by clicking on

The details of an Agent are shown by clicking directly in the table row that distinguishes the agent itself.

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Agents

Create a new AgentTo create an Agent click on the button and fill the form with the following informations:

Username: the Username of the Agent that you want to create, necessary for the Agent’s loginFirst Name: Agent’s First NameLast Name: Agent’s Last NameEmail: Agent’s EmailPhone: Agent’s PhoneMobile: Agent’s MobileAddress: Agent’s AddressZip Code: Agent’s Zip CodeCity: Agent’s CityPassword: a password, necessary for the Agent’s loginProfile Photo: Agent’s gravatarModule Permissions: which modules are visible to the agentSIP Name: the SIP can be associated to the Agent choosing one SIP that already exists in the dropdown menu. If it’s not specified, xCALLY-Shuttle automatically creates and associates also a SIP Agent with the same credentials and default parameters, when a new agent is created.

By clicking on the button it can be specified: - the name of the SIP - the SIP password associated with the Agent - the type of SIP, Web or not.

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Agents

Edit an AgentAs said before, the informations about an Agent can be viewed by clicking on the table row which contains his name.Agent’s details are on the left section, while the informations about the associated SIP are on the right, like shown in this screenshot. For both of them it’s possible to edit the parameters by clicking on or

Another important feature offered by this section is the Queue Association (a Queue can be created and configured in the section “Queues”) to the Agent itself.To associate an Agent to a Queue is necessary to:1. Select one Queue from the list of the existing Queues2. Click the button with the right arrow on it to associate the Agent with the selected queue3. The queue’s name is on the list of the queues currently associated with the Agent4. Click on Save to confirm the associations Queue/Agent

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3-way Agent Experience

Agents can login into the system and use xCally Shuttle in three different ways, thanks to:

1. the powerful CTI Windows phone bar.

2. the Shuttle Web interface and an associated external SIP standard Phone.http://IP-Shuttle or https://IP-Shuttle

3. the experimental Beta version of WebRTC client: the Agent can login inside the Shuttle Web interface with Google Chrome, if a supervisor has created the agent as a Web user (WebRTC is still an ongoing standard, thus you can use it at your own risk).

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3-way Agent Experience

1. The CTI Windows phone barAfter the installation of the CTI phone bar (http://www.xcally.com/bar/co/3/0/17/) the Agent can simply login using his credentials.

Dial the extension 600 to listen to the echo test

Here you can find the full guide http://www.xcally.com/shuttle/xCALLY-PhoneBAR-Guide.pdf and video https://www.youtube.com/watch?v=1SYi7pRqoBA about the CTI Windows Phone bar.

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3-way Agent Experience

2. External PhonesAgents can also use external IP phones. It’s possible to configure this mode through the following steps:

1. The supervisor has to associate the Agent to the proper SIP username related to the external SIP phone.Example: the supervisor has created the agent emily.brown associated to the external SIP phone corresponding to the username digium301.

2. Login using the Shuttle web interface. For example, if your xCALLY Shuttle server has been installed on 192.168.1.10, the agent emily.brown will need to browse and login here:

https://192.168.1.10login: emily.brownpwd: her password

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3-way Agent Experience

After performing the login, the agent will get the powerful Web Workspace console, where he/she can:

- manage the Pause status - see his/her associated Queues - see the other available agents.

Important: The agent MUST keeps logged on to the browser until he/she wants be able to use the external IP phone to manage the customer care calls.

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3-way Agent Experience

3. WebRTCWebRTC is a real time communication standard still under development. It is supported the best on the Google Chrome browser, however it is still under standardization and development: therefore, use such Beta feature at your own risk (no warranty or any guaranteed service)!Here you can find the Google Chrome recommended version:chromium MAC https://provisioning.xcally.com/files/packages/chromium/mac/chrome-mac.zip

chromium LINUX https://provisioning.xcally.com/files/packages/chromium/linux/chrome-linux.zip

chromium WIN32 https://provisioning.xcally.com/files/packages/chromium/windows/chrome-win32.zip

The agent who wants to use the WebRTC interface needs to:1. Be enabled by the supervisor under the SIP section (Web User flag enabled).

When an agent is associated to a SIP Web enabled user he/she can work ONLY with WebRTC (xCALLY Windows phone bar or external phones will not work).

2. Login using the Shuttle web interface using ONLY the Google Chrome recommended version via https (DO NOT use http). For example, if your xCALLY Shuttle server has been installed on 192.168.1.10, the agent emily.brown will need to browse and login here:

https://192.168.1.10login: emily.brownpwd: her password

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3-way Agent Experience

After performing the login, the agent will get the powerful Web Workspace console, where he/she can:

- manage the Pause status - see his/her associated Queues - see the other available agents.

Important: In order to be READY to manage the calls, the agent MUST click on the HEADSETS icon and on the OFFLINE button.

The Agent Shuttle WebRTC workspace is now ready to be used!

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SIPxCALLY Shuttle provides a dedicated SIP section to: - view extisting Asterisk SIP - add new ones - edit parameters available on Asterisk.

In this screenshot is shown the SIP section, with the list of the SIP already created and their main informations, in particular: - Type: SIP User or Web User - Status: Connected or Disconnected

There are two actions that can be done quickly through the SIP table: - Delete the SIP with a click on the button - Change the parameters specified in the creation of a SIP by clicking on

The details of a specific SIP are shown by clicking directly in the table row that distinguishes the SIP itself, where it’s possible to configure advanced parameters offered by Asterisk.

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SIP

Create a new SIPTo create a SIP click on the button and fill the form with the following informations:

Username: the name of the SIP that is being created, it can be either numeric or textual.Password: authentication password of the SIPConfirm Password: the confirmation of the authentication password Caller-ID: defines the identifier, when there are no other information availableContext: the context of the dialplanType: the SIP type can be: - User: used to authenticate incoming - Peer: for outgoing calls - Friend: covers both characteristics of the above.DTMF mode: how DTMF (Dual-Tone Multi-Frequency) are sent: - RFC2833: the default mode, the DTMF are sent with RTP but outside the audio stream. - INBAND: The DTMF is sent in audio stream of the current conversation, becoming audible from the speakers. Requires a high CPU load. - INFO: Although this method is very reliable, it is not supported by all PBX devices and many SIP Trunk.Nat: this parameter specifies that the SIP device is behind a NAT (Network Address Translator) Qualifiy: to determine when the SIP is achievableAllow Codec: with this parameter you can specify the codecs enabled for SIP; some of them are pre-selected by defaultUser Web: Enabling this switch specifies that this will be used with a SIP WebRTC interface (for example the one offered in the xCALLY Shuttle Agent’s view)Description: It ‘s the description associated with the SIP, useful to describe in words his context.

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SIP

Edit a SIPThe informations about a SIP can be viewed by clicking on the table row which contains his SIP internal.In this detailed view you can manage: - General Settings - Web Settings - Advanced Settings.

In the General Settings section it’s possible to edit the informations set during the creation of the SIP, described in the previous page.

The Web Settings represent the WebRTC parameters: - ICE Support: to enable the protocol ICE (Interactive Connectivity Check), which is necessary for the proper functioning of WebRTC. - Has SIP: it specifies the use of the SIP protocol. - Has IAX: it specifies the use of the IAX protocol for the SIP. - Encryption: it points out the need to encrypt the content of calls, thanks to the SRTP (Secure Real-Time Protocol).

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SIP

- Avpf: it enables the use of Audio-Video Profile, which is also required by the WebRTC technology - Video Support: the parameter required for video calls - Description : the description associated with the SIP, to describe in words his context.

In the Advanced Settings you can define the parameters of the SIP to set special features highly demanded by advanced users: - Limit on Peers: to define the call limits of a “peers” SIP type. This can improve the status notification if you are using a “friend” SIP type for incoming calls. It also allows to evalutate incoming and outgoing calls; if it is set to yes value, Aste risk uses a counter for both incoming calls and outgoing. - Call Counter: If enabled, this parameter allows Asterisk to provide useful information about the status of SIP devices. - Can reinvite: thanks to this parameter two devices can establish directly the SIP RTP connection (Real Time Protocol). The result is to minimize the use of resources needed to establish the full-duplex communication. - Direct Media: enabling this parameter, Asterisk tries to drive traffic between the caller and the callee. Not all devices support this feature. -Amaflags: AMA (Automated Message Accounting) allows to classify the SIP calls in the CDR for billing of calls: - Default: sets the default system with the value ‘3’ in the CDR - Omit: does not record calls (setting ‘1’) - Billing: make calls setting ‘2‘ in the CDR and classifying them as billable - Documentation: classifies the call as documentation, setting ‘3’ in the CDR - Subscribe Context: this parameter allows you to specify a context for subscriptions. If it is not set, is the same of that specified in the parameter “context”. - Busy Level: this numeric parameter specifies the number of calls in progress in order to categorize the SIP device as busy. - RTP Timeout: it allows you to automatically terminate the call if you don’t detect RTP traffic within the specified number of seconds. - RTP Hold Timeout: this parameter is more restrictive than the previous one because, unlike RTP Timeout, if the call is put on hold (pause), at the end of this time, the call is terminated. - Set var: here it’s possible to set a variable with its value (using this format: “variable_name=value”). You have to specify that the variable is not case sensitive.

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QueuesxCALLY Shuttle provides also a section dedicated to the Queues, useful to: - create or remove Queues - view the existing Queues and edit their parameters.

In this screenshot is shown the list of the Queues already created and their main informations, in particular the type of strategy chosen to distribute calls.

There are two actions that can be done quickly through the SIP table: - Delete the queue with a click on the button - Change the parameters specified in the creation of the queue by clicking on

The details of a specific Queue are shown by clicking directly in the table row that distinguishes the queue of interest.

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Queues

Create a new QueueTo create a Queue click on the button and fill the form with the following informations:

Queue Name: the name of the queueNext agent call: you can define a distribution policy for inbound calls, selecting one of the following strategies: - Ringall: it contacts all Agents until one answers - RRMemory: one of the most used round robin memory. It resumes from the agent succeeding the one contacted during last call - Least recent: : it contacts the Agent that has recently been less involved in the queue - Fewest calls: it contacts the Agent who has recently answered to the minimum number of calls in the queue - Random: it randomly routes calls to agents - Linear: it contacts the agents according to their order of login - Wrandom: it randomly routes calls to agents using the penalty parameter as weight Description.

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Queues

Edit a QueueThe informations about a Queue can be viewed by clicking on the table row which contains the queue of interestIn this detailed view you can manage: - General Settings - Announce Settings - Advanced Settings - Agents-Queue Association.

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Queues

In the General Settings section you can define the following parameters: - Music on Hold: it allows the choice of different tunes (previously loaded in the audio section) to entertain the caller during the waiting time. The default one is loaded into the Music on Hold Asterisk configurations. - Agent Timeout: the time interval (in seconds) over which the agent is considered unavailable to take charge of a call. - Capacity (Max Len): it specifies the maximum number of calls on hold. If equal to 0, this number is unlimited. - Retry: time interval (in seconds) expected from the system before re-contact all the agents in the queue in object. - Wrapup Time: minimum time interval (in seconds) to contact an agent between two different calls. - Weight: it defines a priority queue. If the agents are multi-skilled, inbound calls will be routed preferentially to the queue with the highest priority. - Join Empty: this setting check if callers can join an empty queue empty, without available agents. In this case there are possible choices: - yes: callers can enter on an empty queue or on a queue where agents are not available - no: callers can’t get on an empty queue - strict: callers can’t enter on an empty queue or on a queue where agents are not available - loose: as strict, but the agents on pause on a queue are considered as available - paused: an agent is not considered available when paused - penalty: an agent is not considered available if his penalty is less than QUEUE_ MAX_PENALTY - in use: an agent is not considered available when he is calling - ringing: an agent is not considered available if his phone is ringing - unavailable: if the agent is part of the queue in question but it is not logged in then it is not considered to be available - invalid: an agent is not considered available if the device status is “invalid” - unknown: it not consideres a member as available if it is unable to determine the current state of the device agents - wrapup: an agent is not considered available if it is currently in its wrapuptime after a call. - Leave When Empty: it works on calls already in the queue - yes: if the queue is empty or agents are not available, the caller leaves the queue. - no: callers remain queued even if the queue is empty or agents are not available. - Calling Record: if active, it records the conversations (by default is disabled).

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Queues

In Announce Settings you can definethe following parameters: - Announce: you can choose different tunes (previously loaded in the Audio section) that will be heard by the agent before talking with the customer. An agent that operates in multiple queues is able to quickly understand what kind of call is and answer appropriately. - Announce Frequency: it sets how often (in seconds) a caller will hear an announcement that indicates its position in the queue or the estimated waiting time. If it’s equal to 0, this function is disabled. - Announce Round (seconds): it’s the level of rounding for the announcements of waiting times. If 0: minutes are considered and announced, without seconds. The possible values are 0, 1, 5, 10, 15, 20, 30. For example, when the value is set to 30, the waiting time 2:34 will be rounded to 2:30. - Announce Hold Time: it indicates whether the estimated wait time will be announced after the information of the position in the queue. The possible values are “yes”, “no” or “once”. - Periodic Announce: you can choose different tunes (previously loaded in the Audio section) to be played to the caller periodically. - Periodic Announce Frequency: it sets how often (in seconds) a caller will hear a periodic announcement.

In the Advanced Settings you can define the following parameters: - Auto Pause (AutoPause): it indicates if an Agent that wasn’t able to automatically answer is paused or not. - Ring in Use: it indicates that queue’s calls have to be forwarded to the agents which have devices in use. - Member Delay: it indicates the interval of silence (in seconds) that the caller hears before being connected to an agent. - Timeout restart: it indicates if the response timeout of an agent is reset after a busy signal or congestion. This can be useful for agents who are authorized to refuse calls. - Service Level: it sets the threshold of service level; for example it sets the maximum wait time for callers. This is very useful for statistical analysis (for example to answer the question “How many calls were answered within the service level threshold for x seconds?”). - Monitor Join: it combines files -in and -out of the audio recordings in a single file. Values: yes or no (default). - Events when Called: it sends events to the Asterisk Manager about the states of the

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Queues

agents before, during and after the call. - Events member status: it sends events to the Asterisk Manager about the states of the agents in the queue (default “yes”). - Report Hold Time: it indicates the possibility to announce the caller’s waiting time to the agent. The possible values are “yes” and “no”. - Context: it indicates the context in which it’s passed the call if the caller presses a key combination while waiting in the queue. This combination is considered as an extension of Asterisk and the call is then removed from the queue and routed to the internal in this context.

In the Agents-Queue Association section the selected Queue can be associated with multiple agents. Remember to click the button “Save” after the association.

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TrunkThis xCally Shuttle Section is dedicated to the managing of Trunks, useful for the creation of the inbound external rules and outbound rules.

In this screenshot there is the list of the already created Trunks with their main informations, like the Host IP and the status (connected or disconnected), which will be shown as Disconnected until you are able to perform a successful inbound call.

There are two actions that can be done quickly through the Trunk table: - Delete the trunk with a click on the button - Change the parameters specified in the creation of the trunk by clicking on

The details of a specific Trunk are shown by clicking directly in the table row that distinguishes the trunk of interest.

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Trunk

Create a new TrunkTo create a Trunk click on the button and fill the form with the following data:

Important: remember that the Host must be an IP Address and that the Registry field must be compiled in this way: username:password@sip_server

Please consider the Trunk Status will be shown as Disconnected until you are able to perform a successful inbound call!

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Trunk

Examples

Scenario 1

Example 1

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Trunk

Example 2

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Trunk

Scenario 2

Example 1 - Gateway Digium E1

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Trunk

Example 2 - Gateway Inalp-Patton

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Dial PlanThe Dial Plan module is dedicated to the managing of the logical routing of communications.This section allows to define rules for the outbound and inbound calls routing. In addition, for each rule, it’s possible to define a specific behavior according to different moments of the day or of the week.

In this screenshot there is the list of the Inbound Rules with the main informations about the type of rule (Internal or External) and the DID.

To show and manage the Outbound Rules you can click on this switch:

There are two actions that can be done quickly through the Rules table: - Delete the rule with a click on the button - Change the parameters specified in the creation of the rule by clicking on

The details of a specific Rule are shown by clicking directly in the table row that distinguishes the rule of interest.

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Dial Plan

Create a new Inbound RuleTo create an Inbound Rule click on the button and fill the form with the following informations:

DID: geographic number or internal numberingNameType: it indicates the rule’s type: - Internal: used for internal numbers. - External: used for geographic numbers. Description.

In addition to the creation form it is necessary to specify other rule settings, as described on the next page.

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Dial Plan

Edit an Inbound RuleThe informations about an Inbound Rule can be viewed by clicking on the corrisponding table row. In the Advanced Parameters you can add applications to the rule in question.For each application, it’s possible to: - Edit it, clicking on - Remove it, clicking on - Set its priority over other applications, clicking on - Set when the application is valid, clicking on .

There are different types of applications, which can be associated to the rule thanks to these buttons:

1. Application IVR Cally-SquareIt refers to a Cally Square project (previously loaded in the Cally Square section).

2. Application AGI (http://www.voip-info.org/wiki/view/Asterisk+cmd+AGI) - Agi Script: set the AGI’s name (Ex. agi :/ / 127.0.0.1/square, project = 3) - Agi parameter 1 .. 6: Possible parameters to AGI.

3. Application Dial (http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial) - Type/ID: phone call (eg SIP/1003) - Timeout: Timeout before the application terminates - Options: Refer to the Asterisk’s wiki - URL

4. Playback Application - Audio File Path - Options(Skip,NoAnswer,J,Say)

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Dial Plan

5. Queue Application -QueueName -QueueOptions - URL - Announce Override - Timeout - AGI - Macro - Gosub - Rule -Position

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Dial Plan

Create a new Outbound RuleTo create an Inbound Rule click on the button and fill the form with the following informations:

Name: rule’s name.Pattern: it indicates the number to be filtered. It may be a fixed pattern or a pattern which represents a series of numbers. The operators most useful are “X”, which indicates a number from 0 to 9, and “.”, which indicates 0 or more numbers from 0 to 9. So, for example, “3X” means from 30 to 39 and “12X.” means all numbers starting with 12 plus at least one other number, like 124, 120, but not only 12)Prefix: the prefix which is necessary to consider before dialing the call number.Cut Digits: the digits that should to be cut to the number before passing the call to the provider.Campaign Log: it indicates the possibility to log all outgoing calls from the rule in question with a campaign name (previously configured in the Settings section).Recording Call: it indicates the possibility to record outbound calls on the rule in question (in WAV or GSM format).Description.

In addition to the creation form it is necessary to specify other rule settings, as described on the next page.

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Dial Plan

Edit an Outbound RuleThe informations about an Outbound Rule can be viewed by clicking on the corrisponding table row. In the Advanced Parameters you can add routes to the rule in question.For each route, it’s possible to: - Set its priority over other routes, clicking on - Set when the route is valid, clicking on - Set Dial options, clicking on - Remove it, clicking on - Edit it, clicking on .

To add a route click on the button and then fill the form with the following parameters: - Suffix: it is required by the provider before passing the call to the provider itself. - Trunk: the output channel (which is previously configured in the Trunk section) - CallerID: ID of the caller. In this case it’s the number to show to the called person. - Context: the context name on which the rule runs.

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AudioThe Audio menu is dedicated to the upload and the managing of different types of audio files, collected in three different sections: .

The Sounds Section for generic audio files, the Music on Hold Section for audio files that you can use during the caller’s waiting time, the Monitor Section for the recorded calls.

There are some actions that can be done quickly through the Audio table: - Add a new Sound or a new Music on Hold , filling this form:

- Delete the audio file - Edit the audio file parameters - Download a copy of the audio file - Listen to the audio file preview

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Cally SquareThe Cally Square menu contains the tool to create and manage IVR applications for your Asterisk based telephony system.

In this section is shown the list of the Cally Square IVR applications already created.

There are some actions that can be done quickly through the Cally Square table: - Delete the IVR application project - Edit the IVR application project parameters - Edit the IVR application project structure and design - Clone the IVR application project

The details of a specific IVR application project are shown by clicking directly in the table row that distinguishes the IVR application of interest.

Each time you get a new IVR channel license you need to login inside the Linux server via ssh and launch the following command to restart the service: service agisquare restart

You can find the full documentation of Cally Square at this link: http://www.callysquare.com/documentation/

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Pre AnalyticsxCally Shuttle includes also the Pre Analytics menu dedicated to reporting, divided into these sections:

The first section contains a list of Call Detail Records with their corresponding data fields about starting time of the call (date), caller informations, source and destination numbers, channel type, call duration, billed seconds, disposition (answered or not), call ID and campaign.

The Inbound Section shows the list of the inbound calls and their informations about starting time (date), agent’s username, caller number and ID, status (answered or not), queue’s name, duration and holdtime, if answered, or waittime if unanswered.

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Pre Analytics

The Outbound Section shows the list of the outbound calls and their informations about starting time (date), agent’s username, destinationand call ID.

The last section is dedicated to the calls routed to IVR applications. Here you can find informations about starting time (date), caller ID, DID, caller choices,IVRblocks,CallIDandCallySquare project’s name.

It’s also possible to copy, print and download all these data through the panel buttons:

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SettingsxCally Shuttle provides a detailed Settings Menu, with all these different sections: - General Settings - License - Asterisk Manager - Integrations - Calendar - Templates - Users - System Updates - Backup - Cronjob - Tickets - Customize - Logs

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Users

This section contains the Users list, to which it’s possible to assign different roles and different accesses to the xCally Shuttle Modules.

The Users table shows informations about their Role, Status (if users are active or not), Last Login date and if the user is an Administrator or not.

Through this table you can edit or remove an existing User or create a new one providing the following data:

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Users

Let’s focus on the main fields: - Username and Password will be used by the User to login into the system - The Title indicates the User role - If you want to create an Admin User you have to select “yes” in the Admin field - In the Module Permissions you can select which of the xCally Shuttle sections can be viewed and managed by the User.

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Zendesk Integration

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xCALLY Shuttle provides a seamless CTI integration with Zendesk, the Cloud Help Desk solution for multi-channel customer cares.

Key benefits are really able to provides your customer care agents a Unified Agent Desktop experience, together with:

- Push CTI technology for automatic Ticket creation each time an Agent receive a call- Trigger configuration to select with Asterisk Shuttle Queue are involved in the automatic Ticket pop-ups and which Events engage it: before answering, after answering or after a call hangup- It WORKS either with EVERY KIND OF SIP PHONES registered on the Asterisk xCally Shuttle PBX, or the xCally phone bar, or the WebRTC client- Great VoIP quality- Great flexibility: use your own SIP trunk with your local providers or on the cloud according to your needs- Totally Asterisk RealTime technology based- Call Recordings option available on the Shuttle and Zendesk web interface- IVR enabled: Cally Square drag and drop IVR totally integrated with Zendesk- Simple and fast xCALLY Shuttle installer- Many other nice benefits included

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Zendesk Integration

In the Setting Menu there is a section dedicated to the integration of xCally Shuttle with third party applications, like Zendesk.

You can find this section by clicking on the Integrations button, like shown below:

In the Integrations page there are two tables: the first contains the list of the Integrations already created and the second one the Triggers list, to determine when and how the integration works.

Create a new IntegrationTo create a new Integration click on the button and fill the form with the following informations:

You can choose a Name which represents the Integration and then fill the Username and Password fields, which require the same username and password of an Administrator Zendesk Account. Then compile the URI field in this way: your Zendesk URL/api/v2/ (for example: https://xcally.zendesk.com/api/v2/) and the Integration Type, in this case Zendesk.

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Zendesk Integration

After the creation of the Integration you can: - check if the data correspond to a Valid Admin Integration Account, clicking on - edit the Integration parameters - delete the Integration

Create a TriggerNow let’s focus on the Triggers Table: here you can set how the integration works by adding a new Trigger. To create it click on and choose your integration strategy filling the form:

Through the Event field you can decide when the integration starts, so when the caller’s ticket will be opened in the Zendesk interface: - Ringing: when the phone is ringing, before the agent’s answer - Up: when the agent picks up the call - Hang up: when the agent hangs up the call.You can also choose for which Queue the integration is valid. If you need to apply the Integration on more queues, you have to create one trigger for each queue.In the last field you can select one of the Integrations you created.

After the creation of the Trigger you can edit parameters or remove it .

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Zendesk Integration

ImportantThe integration between xCally Shuttle and Zendesk uses the new Shuttle Push Technology in order to speed the Ticket creation and agent pop-up.In order to have it working properly you need to be sure that:

1. the Agent Name in the Zendesk profile corresponds exactly to the Agent Caller-ID name in the xCally Shuttle Agent profile (SIP details), like shown in the image below.

2. The Agent is associated to a Group, in the Zendesk Agent Profile. Administrators can create Groups and associate Agents to them.

https://support.zendesk.com/entries/20047256-Creating-managing-and-using-groups

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Zendesk Integration

3. the Agent MUST have a Zendesk role different from Administrator and End User. It must have one of the Zendesk AGENT ROLES.If the Agent is associated with an Administrator or an End User account, he will not be able to see the automatic Ticket creation!

https://support.zendesk.com/entries/20055812-Understanding-Zendesk-user-roles

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Zendesk Integration

HintsxCALLY phone bars: if you use the xCALLY phone bars, you just need to create the Agents, being careful to fill in with the Name same as your Zendesk AGENT NAME. Do not touch the Caller-ID SIP created field (it will be automatically generated with the right value to make the integration working).

External IP phones: if you use external IP phones, or experimental WebRTC, you just need to create the Agents, being careful to fill in with the Name same as your Zendesk AGENT NAME. Afterwards, just link the Agent to the SIP username related to your external IP phone. The Caller-ID SIP field and the entire SIP related account, can be modified later on without any impact.

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Zendesk Integration

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Custom fieldsCustom ticket fields are typically used to gather more information about the support issue or product or service in Zendesk.Each Zendesk custom field is associated to one ID.Using Shuttle you can now create a MAPPING between the Zendesk custom field ID and the available field values.

In order to do it just enter in the Shuttleadministrationweb interface -> Section Integrations and add your custom field mapping in the CUSTOM FIELDS sub-section clicking on .

Fill the form with the requested informations:- Custom Field ID: Zendesk Custom Field ID (the image on the next page indicates how to retrieve it)- Prefix Custom Field: String prefix- Available Fields

- From: Caller - Date: Enter Time Call- To: Called - Unique ID: Asterisk Unique ID Call- Id: Caller ID Zendesk - Recording Link: Link Recorded Call- Name: Caller Name - SIP: Shuttle SIP- Queue: Shuttle Queue - Agent: Shuttle Agent- Position: Queue’s Position- Count: Queue’s Count- Trunk: Shuttle Trunk

- Suffix Custom Field: String Suffix- Integration: Integration Name

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Here it is a mapping example:

Through the Custom Fields table it’s possible to: - delete a custom field, with a click on the button - change the parameters specified in the creation of the custom field, by clicking on

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Here it is the result: your agents will get the ticket pop-up with the custom fields auto-filled.

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TroubleshootingThe Integration doesn’t work? Check carefully the following steps:

1. Click on this button , on the xCally Shuttle web interface, to verify if the integration data you gave are correct.Thefeedbackisnegative? Check if: - the username and password correspond to a valid Zendesk Admin Integration Account - the Zendesk URI is correct.

2. The Agent is correctly associated to the Queue specified in the Trigger that you created (xCally Shuttle - Integration Settings).

3. The Agent Name in the Zendesk profile corresponds exactly to the Agent Name in the xCally Shuttle Agent profile.

4. The Agent is associated to a Group, in the Zendesk Agent Profile.

5. The Agent has a Zendesk Role different from Administrator and End User.

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