Der Call Flow Editor (CFE) ist ein vollständig in die Weboberfläche von AskoziaPBX integriertes Zusatzmodul. Mit Drag-and-Drop können Sie in kürzester Zeit Ihre eigenen Call Flows erstellen und so komplexeste Szenarien realisieren, wie z.B. interaktive Sprachdialogsysteme, Warteschleifen, intelligente Nachtschalter, persönliche Ansagen, Anrufaufzeichnung und vieles mehr.
Mit der Integration von CFE bietet SipXecs/openUC eine super alternative ACD Lösung.
Askozia with the Call Flow Editor (CFE) offers an ACD solution as an alternative to the SipX/openUC integrated ACD solutions.
Main Features of the Askozia ACD system:
This guide will describe a way to connect Askozia as an ACD application system to a SipX/openUC PBX. To have all the features described here you need a SBC (Session Border Controller). The configuration description is based on a Patton Smartnode Gateway that was configured to act as a SBC. This gateway is often used for PSTN connectivity. Thus it is already installed on most systems and can alongside (if no transcoding between codecs) handle the SBC function for the ACD application. Other SBCs may work too (ACME, Karoo, Audiocodes Mediant etc.) but have not been tested yet. They have to be compatible to SipX (PBX side) and Asterisk (ACD application side).
This scenario describes basic configuration of call queues, static agents and dynamic agents including scripts to show the status of acd agents on Polycom phones. It is not intended to be a complete guide. But it should be enough to lead you on the right track.
The following illustration shows how Askozia is connect to the SipX PBX and which ways a call to a queue is taking.
The configuration steps expect that you have a running Askozia box with the call flow editor that is reachable by its IP address or domain. The installation is pretty simple:
It also expects that you have a running SipX / OpenUC installation. The steps to get a running Sipx are described in other wiki articles. It also expects that you have a Patton Smartnode Gateway (for other SBCs you have to adapt the configuration steps).
We will start with the configuration of the SBC because it is a transparent gateway between SipX and Askozia and normaly does not have to be touched again. It will simply route all calls from SipX to Askozia and from Askozia to SipX.It uses two SIP interfaces. One for SIP communication with SipX. The other with SIP communication with Askozia. This examples uses port 5080 to communicate with SipX and 5090 to communicate with Askozia.
context sip-gateway GW_YOURCOMPANY_SBC interface IF_YOURCOMPANY_SBC bind interface WAN context router port 5080 |
location-service LOC_SRV_SIP domain 1 voip.yourcompany.com 5060 identity-group default call outbound use profile tone-set default use profile voip default use profile sip default preferred-transport-protocol udp invite-transaction-timeout 2 non-invite-transaction-timeout 5 call inbound use profile tone-set default use profile voip default use profile sip default |
interface sip IF_YOURCOMPANY_SBC bind context sip-gateway GW_YOURCOMPANY_SBC route call dest-interface IF_ASKOZIA remote voip.yourcompany.com early-disconnect call-reroute accept call-reroute emit address-translation outgoing-call from-header user-part call host-part fix voip.yourcompany.com 5080 address-translation outgoing-call to-header user-part call host-part fix voip.yourcompany.com use profile voip ACD-SBC |
profile voip ACD-SBC codec 1 g711alaw64k rx-length 20 tx-length 20 dtmf-relay rtp flash-hook-relay rtp rtp traffic-class local-default |
context sip-gateway GW_ASKOZIA interface IF_ASKOZIA bind interface WAN context router port 5090 |
location-service LOC_SRV_ASKOZIA domain 1 askoziaIP 5060 identity-group default call outbound use profile tone-set default use profile voip ACD-SBC use profile sip default preferred-transport-protocol udp call inbound |
interface sip IF_ASKOZIA bind context sip-gateway GW_ASKOZIA route call dest-interface IF_YOURCOMPANY_SBC remote askoziaIP 5060 early-disconnect address-translation outgoing-call from-header user-part call host-part fix gateWayIP 5090 address-translation outgoing-call to-header user-part call host-part fix askoziaIP 5060 use profile voip ACD-SBC |
We have to configure four things:
1,Answer() n,GotoIf($[„${DB(AGENTONLINE/${EXTEN})}“ = „1“]?online:offline) n(online),Set(DB(AGENTONLINE/${EXTEN})=0) n,RemoveQueueMember(CALLFLOW-17152790674fec1bc8139a5-QUEUE-3,SIP/401@SIP-PROVIDER-13804556994f58c906819d8) n,Set(DEVICE_STATE(Custom:agent${EXTEN})=NOT_INUSE) n,Playback(de-activated) n,Hangup() n(offline),Set(DB(AGENTONLINE/${EXTEN})=1) n,AddQueueMember(CALLFLOW-17152790674fec1bc8139a5-QUEUE-3,SIP/401@SIP-PROVIDER-13804556994f58c906819d8) n,Set(DEVICE_STATE(Custom:agent${EXTEN})=INUSE) n,Playback(activated) n,Hangup() |
Advanced Configuration
Lets say there is an ACD agent (extension 130) that is not statically configured for a queue. When he starts working, he is logging in using a speeddial and when he stops working he is logging of again. The agent is using a Polycom phone and you want to show if he is logged in on the screen of the phone. For this function you need an http server with php around. You may also install php on your SipX system and use that one.
Place the following php script on your webserver and replace the IP and the password in the URL call to the ACD system. You also have to place two images on the webserver. On that is showing logged out and the other showing logged in. The example script delivers an image suitable for Polycom Soundpoint IP 450 phones.
<?php |
You can test the script using your browser. Replace the id with the login extension you chose for your agent (e.g. 7130 for agent 130).
http ://webserverIP:65080/polycom/polycomACD.php?id=7130 |
If that works you can take step two:
Askozia lets you monitor your call queues. In the callflow editor you can click the wrench of a queue element to access the queue settings. Here is a link to the wallboard that shows the status and statistics of the queue.