L'offre PSTN Calling avance vite en Europe, c'est au tour de l'Allemagne et la Belgique d'être disponible en mode preview:
Inscription ici: https://www.skypepreview.com/
L'offre PSTN Calling avance vite en Europe, c'est au tour de l'Allemagne et la Belgique d'être disponible en mode preview:
Inscription ici: https://www.skypepreview.com/
Un nouveau programme de preview est disponible en relation avec des fonctionnalités a venir concernant le PSTN Conferencing.
Description ici: https://techcommunity.microsoft.com/t5/Skype-for-Business-Blog/Skype-for-Business-New-PSTN-Conferencing-Enhancement-Features-to/ba-p/76494
providing administrators with additional user-level controls for toll-free and dial-out destinations for Skype for Business meetings.
Control which users can leverage toll-free numbers
This feature will allow tenant administrators to enable or disable the usage of toll-free numbers for the meetings of any organizer on a per user basis.
When toll-free numbers are disabled for a given user:
Restrict dialing out in meetings for specific users
This feature will allow a tenant administrator to control which users can dial out from within a meeting.
This would include:
Inscription: https://www.skypepreview.com/
Si vous faîtes parti du programme Office Insider, une mise a jour du client Skype Entreprise MAC est disponible depuis le 8 juin comportant les nouveautés suivantes:
Version: Build 16.8.0.170
Features added
Users can now join non-federated meetings directly from Skype for Business, rather than having to join in their browser.
New Contacts tab shows who has added you to their contacts list in Skype for Business.
Users can now join a Microsoft Teams meeting directly from the Meetings tab in Skype for Business .
Users can close the "On Hold" overlay while in a meeting. Held calls can be resumed from the new More Options button (...).
Users can view screen-sharing or PowerPoint presentations side-by-side with the active speaker.
Touch bar support for various functionality.
Improvements
Fixed a mute/unmute problem that would happen with various headsets.
Fixed a case where the microphone icon would look wrong when joining a conference.
Implemented minor bug fixes.
Intéressante annonce, les ports 50,000-59,999 UDP and TCP deviennent optionnels dans le cadre d'un projet avec du Skype for Business Online.
Enfin, les Polycom RealPresence Group Series (310,500,700) sont officiellement supportés dans Office 365 Skype Online depuis le 1ier juin 2017.
Version supportée: Polycom Group Series VTC version 6.1.1
Le 1ier juin, Sonus a rendu disponible la version SBC SWe Lite 6.1.2.
Version: Sonus SBC SWe Lite Release 6.1.2 build 104
Nouveautés:
Corrections:
| CHOR-189 | Music On Hold Support. |
|---|---|
| CHOR-1015 | ESXi will not come up after setting a static MAC address. |
| CHOR-1057 | System reboot pushes the SWE Lite clock ahead. |
| CHOR-1112 | SCOM Support. |
| CHOR-1119 | Raise alarm if countable license is beyond SWe Lite capacity. |
| CHOR-1122 | LM does not set expiration date for BRSF license. |
| CHOR-1124 | Hyper-V instance cannot handle 100 concurrent calls with complex codec. |
| CHOR-1150 | Add Q.850 to SIP and SIP to Q.850 mapping |
| CHOR-1159 | Handle ROC for non-SFB endpoints. |
| CHOR-1160 | SWe Lite should have unique sysObjectID. |
| CHOR-1171 | Calls failed with G722 SS enabled. |
| CHOR-1172 | Media System Control stability improvements. |
| CHOR-1173 | netconfig user prompts whether to configure static IP after initial setup. |
| CHOR-1214 | double reboot for KVM and ESXi after installation. |
| CHOR-1276 | One way audio in SIP to SFB calls. |
| SYM-22988 | SBC accepted register request after call route failure in Access Mode. |
| SYM-22989 | SIP stability improvements. |
| SYM-22990 | Cause Code Mapping table does not support several SIP responses. |
| SYM-23027 | INVITE without SDP and PRACK with SDP handling. |
| SYM-23033 | UX: SBC2000 sends "486 Busy Here" when a phone tries to pick up the parked call for second time. |
| SYM-23048 | SIP message handling improvements. |
| SYM-23056 | Radius stability improvements. |
| SYM-23162 | The password plugin shows the password in plain text when the field display is toggled. |
| SYM-23165 | SBC doesn't response for 401 Unauthorized challenge if outbound proxy is configured. |
| SYM-23209 | Add AD Cache Refresh action to REST. |
Problèmes connus:
| CHOR-1067 |
Multiple codecs, with SRTP, and media bypass Disabled, causes intermittent transferred call drop. |
Available workarounds:
|
Lien: https://support.sonus.net/display/UXDOC61/SBC+SWe+Lite+6.1.2+Release+Notes
Contexte:
Petit problème du jour sur mon propre poste de travail. Aucun changement, j'étais en maintenance sur un environnement Cisco la veille au soir, aucun problème, j'ouvre mon poste de travail ce matin, le client Skype Entreprise n'est pas capable de s'exécuter.
Résolution:
1/ Lancer un command prompt avec des droits d'administrateurs.
2/ Lancer l'outil SFC
sfc /scannow
3/ Redémarrer votre poste de travail a la fin du scan.
4/ Le client Skype fonctionne!
Le KB 4023993 est a prendre en considération si vous êtes dans le scénario suivant:
You deploy Microsoft Lync Server 2010, Microsoft Lync Server 2013, or Microsoft Skype for Business Server 2015.
The Microsoft .NET Framework 4.5.2 or a later version is installed (Lync Server 2013 or Skype for Business Server 2015).
You install the May 2017 the .NET Framework Security and Quality Rollup.
Expériences vécu dans le cadre de ces scénarios:
The Lync Server 2010, Lync Server 2013, or Skype for Business Server 2015 Front End server generates the following LS Data MCU event 41026 error.
Il existe plusieurs options possible afin de revenir a une configuration d'usine pour une passerelle Audiocodes MSBR 800.
1/ CLI
Prérequis:
Au prompt, préciser votre username et password, puis faire les commandes suivantes.
enable
write factory
2/ Fichier ini
Prérequis:
Importer un fichier ini vide.
3/ Hardware Reset
Rester appuyer entre 12 (minimum) et 24 (maximum) secondes sur le bouton de reset.
Observer sur plusieurs déploiements différents, la mise a jour Skype Entreprise CU5 de Mai 2017 génere une erreur sur les serveurs Frontend de type LS Data MCU
Contenu:
Merci a ce blog pour la résolution!
Pour ma part, j'ai choisi l'option de regénérer les certificats internes des Edge avec l'authentification Client et Serveur dans le EKU
C'est officielle, depuis le CU5 de Mai 2017, Windows Server 2016 est supporté pour Skype for Business 2015.
Information ici: https://support.microsoft.com/en-gb/help/4015888/supports-to-use-windows-2016-as-the-operating-system-in-skype-for
Depuis la mise a jour Skype Entreprise 2015 CU5 du mois de mai 2015, il est possible d'activer Skype Meetings App afin d'améliorer l'expérience lorsqu'un participant rejoint une conférence.
C'est quoi Skype Meetings App? description ici: https://support.office.com/en-us/article/What-is-Skype-Meetings-App-Skype-for-Business-Web-App-1ff3d412-718a-4982-8ff2-a4992608cdb5
Description Skype Entreprise 2015 CU5: http://microsofttouch.fr/default/b/christophe/posts/skype-for-business-cumulative-update-skype-for-business---mai-2017--
Avant de procéder aux modifications de configurations, les valeurs sont les suivantes:
L'interface est Skype Web App lorsqu'un participant rejoint une conférence.
Étapes d'activations:
1/ When you enable access to the Content Delivery Network (CDN), users will have the ability to connect to CDN online and get Application Réunions Skype, and will use the simplified meeting join experience.
Set-CsWebServiceConfiguration -MeetingUxUseCdn $True
2/ Allow client side logging telemetry from the meeting join web page or the Application Réunions Skype to be sent to Microsoft servers (the command defaults to false).
Set-CsWebServiceConfiguration -MeetingUxEnableTelemetry $True
3/ Set the timeout before fall back to the locally hosted Application web Skype Entreprise experience if CDN isn't available. The default value is 6 seconds. If this value is set to 0, there will be no timeout.
Set-CsWebServiceConfiguration -JoinLauncherCdnTimeout (New-TimeSpan -Seconds 10)
4/ Lancer la commande: Enable-cscomputer
Valeurs suites aux changements de configurations:
L'invitation de réunion est la même, l'expérience utilisateur est différente suite aux modifications des configurations.
Informations:
KB Update that enables simplified meeting join experience in Skype for Business Server 2015: https://support.microsoft.com/en-us/help/4015907/update-that-enables-simplified-meeting-join-experience-in-skype-for-business-server-2015
Skype Meetings App: https://support.office.com/en-us/article/Skype-Meetings-App-help-Skype-for-Business-Web-App-e08370be-2fbb-4ce9-9a90-c84d92cc4cab?ui=en-US&rs=en-US&ad=US
End-user help topic: Skype Meetings App help (Skype for Business Web App)
Planning topic: Plan for Web downloadable clients
Deployment topic: Deploy Web downloadable clients in Skype for Business Server 2015
Contexte:
Dans le cadre d'un projet de migration Asterisk vers Microsoft Skype Entreprise, afin d'assurer une coexistance avec migration progressive des 50 sites, les 3 PRI sont déplacés du Cisco ISR 2921 vers une passerelle Audiocodes Médiant 1000. Ce dernier est responsable du routage des DID en fonction de l'avancée de la migration. Afin de ne pas avoir a modifié les configurations du systeme Asterisk, le routeur Cisco est conservé temporairement de l'infrastructure et connecté derrière l'Audiocodes afin d'assurer le relais avec l'ancien environnement ipbx.
Probleme:
Suite au déplacement des PRI vers la passerelle Audiocodes, pendant la phase de test, on se rend compte que les options ne sont pas pris en considérations lors d'un appel vers une des réceptions automatisées ou IVR gérés par l'environnement Asterisk.
En regardant les traces, on observe que la passerelle Audiocodes propose dans la négociation d'échanger des tonalités DTMF au format RFC 2833 comme attendu.
Du côté du routeur Cisco, le incoming dial-peer est aussi configuré pour négocier et supporter les codecs G.711 et les tonalités DTMF au format RFC 2833.
Mais malgré cette configuration, on observe dans la réponse du routeur l'absence de la ligne:
a=rtpmap:101 telephone-event/8000
Résolution:
La passerelle Audiocodes par défaut est configuré avec un payload de 96 pour le DTMF, les routeurs Cisco ISR avec un payload de 101.
Deux options pour faire fonctionner l'intégration.
Changer au niveau du routeur Audiocodes pour s'ajuster avec les attentes Cisco:
Changement au niveau du dial-peer du routeur Cisco pour s'ajuster avec les attentes Audiocodes:
Router(config-dial-peer)# rtp payload-type nte 96
C'est officiel, les firmwares en version 3.0 sont disponible pour les téléphones IP Audiocodes série HD400.
La premiere version GA est 3.0.575.42.
Les modèles compatibles sont les HD450, 420, 440 et 450.
Les nouveautés:
- Online sign-in – connectivity to Office 365. New capability to sign in to (connect to) and authenticate with Microsoft's Cloud PBX, Microsoft's cloud-hosted version of enterprise voice. AudioCodes' phone features two new sign-in method options, allowing users to connect to Microsoft's Cloud PBX:
• ADAL (Azure AD Authentication Library) that is based on OAuth 2.0 (RFC 6749). The phone always starts with ADAL and if it's unavailable on the server side, the phone moves to OrgID.
• OrgID (Organizational ID) or LiveID is Microsoft's proprietary connectivity to Cloud services.
- Multi-Party Skype for Business Remote Conferencing utilizing CCCP (Centralized Conference Control Protocol) is now supported on the phone. A new Meet Now/Conf softkey is displayed by default in the 400HD phones. The softkey allows users to easily initiate remote multi-party Skype for Business conference calls. By pressing the new softkey, users can initiate, join or be added to a multi-party conference call while having full control and viewing capability.
Users can now:
• View the Roster – see other participants and their status (like the Mute option, Hold status)
• View the conference PSTN dial-in number and conference ID
• Mute/Unmute other participants
• Manage the conference status as Lock/Unlock
• Manage the Lobby for Conference calls that Lobby is defined – Admit/Deny other participants
• Presenters in a conference can add users to the conference
• Presenters in a conference can remove users from the conference
• Presenters in a conference can change the role of a participant between 'presenter and attendant'
In versions prior to Version 3.0.0.575.42, supported conference capability was locally based (phone based) and limited to three users in a 3-way conference, or remote based, with more than two parties from the Skype for Business client, using the BToE feature.
• Merging a call into a conference. Two separate calls can now be merged into one
conference call. This can be performed via a new Merge option accessed from the phone's
Call Menu softkey, or via the Skype for Business client if the user is paired.
Integration with Microsoft Exchange Server (Calendar) + click to join a Skype for Business meeting. [Applies to all AudioCodes phones except the 420HD]. Users can view their Microsoft Exchange Calendar meetings in the phone's LCD by selecting a new Calendar option from the MENU key. The phone by default displays meetings scheduled to commence between the present and 24 hours from the present (24H), but the network administrator can change the default and configure the phone to display meetings scheduled to commence between the midnight of the night before the present and the midnight of the night ahead (TODAY). Via the phone, the user can join any online meeting scheduled in Skype for Business: A Join softkey is displayed for the user to join 400HD Series IP Phones for Skype for Business in the meeting online. To connect to Microsoft Exchange and receive the Calendar feature, sign-in must be with username in UPN format, as described in the Note above.
Visual Voice Mail. [Applies to all phones except 420HD]. By pressing the voicemail key on the phone, users can now see a list of voicemail messages and select which message to listen to or delete. The user’s voicemail must be enabled to allow this feature. When a call comes in, the caller can be referred directly to voicemail by pressing a To VM softkey displayed when the phone rings
BToE
• Automatic Pairing (requires BToE PC/laptop application Version 2.x). Users no longer need to manually pair the BToE PC/laptop application with the phone. If the laptop after automatic pairing is disconnected and moved to another location, its speaker/headset becomes the audio device associated with the Skype for Business client.
If the laptop is manually paired and then relocated (manual pairing is still an option), Skype for Business audio remains through the phone. It's therefore advisable to pair automatically.
Note: If BToE with manual pairing has already been performed on a PC/laptop and you want to automatically pair, you must delete the old pair code from the BToE PC/laptop application in order to allow BToE automatic pairing.
• Support for video calls. When a video call comes in, video is displayed on the PC/laptop, voice is routed to the phone. By Skype for Business design, the phone performs similarly to a
USB device during this scenario. The feature is supported only if the user's phone was automatically paired (by connecting its PC port to the PC/laptop ‘behind’ it).
• Switching between audio devices. Users can switch back and forth between audio devices. A user in an active call can switch from the phone to using a USB headset connected to the PC, for example, and then back to using the phone. In this scenario, when going back to using the phone, the phone performs similarly to a USB device (by Skype for Business design). The feature is supported only if the user's phone was automatically paired (by connecting its PC port to the PC/laptop ‘behind’ it).
Phone Automatic lock. The Skype for Business phone now supports the capability to automatically lock after a preconfigured period of time. The feature secures phones against unwanted (mis)use. When the phone is locked:
• Incoming calls are allowed but outgoing calls require a security PIN code
• Without the PIN code, the Call Log, Calendar and Corporate Directory
Locking / unlocking a paired phone: If a user's phone was automatically paired (by connecting its PC port to the PC/laptop ‘behind’ it) and if the PC/laptop is active (not locked), the phone cannot be manually locked. The user can manually lock it only after locking the PC/laptop. If the user doesn't manually lock the phone, it will nevertheless automatically lock after the timeout preconfigured in the Skype for Business server lapses. The phone will unlock only after the user unlocks their PC/laptop or if the user manually unlocks the phone.
Capability to handle multiple calls - N Concurrent calls (NCC). The phone is capable of managing up to 8 concurrent calls per line, for example, of holding multiple calls and switching
between them (most relevant to the receptionist)
Incoming calls to a Delegated Line are displayed in the sidecar of the 440HD IP phone (exclusively). A new option was added to display in the phone's sidecar incoming and outgoing
calls which users can pick up. The phone is capable of presenting up to 12 active calls (limited by the number of the phone's sidecar keys), and of handling up to eight calls simultaneously.
Integrated Log Upload. Allows uploading logs from the phone to the Microsoft server for troubleshooting/support purposes. Complies with Microsoft's certification requirements for 3rd party Skype for Business clients.
Device Update. The Skype for Business server can update the IP phone firmware version. For detailed information on the update process, refer to https://technet.microsoft.com/enus/
library/gg398861.aspx/
Quality of Experience (QoE) reports are now sent to Microsoft's SQL server. The phone supports QoE reporting directly to the Skype for Business monitoring tool. Supported metrics include the voice quality parameters of Jitter and Packet Loss.
Skype for Business 'Favorites' contacts & Outlook contacts integrated with the phone.
[Applies to all AudioCodes phones except the 420HD). Contact groups defined in Skype for Business & Outlook contacts are now integrated with the phone. Pressing the CONTACTS hard key on the phone displays by default the 'Favorites' contact group defined in the Skype for Business client. The user can dial a contact directly from it. In addition, pressing the Menu softkey in the Favorites' screen provides the option to access other 'Contact groups' such as 'Outlook Contacts' or "Family'.
New codecs supported: Skype's SILK 8000 and SILK 16000. SILK is an audio compression format and audio codec that can use a sampling frequency of 8, 12, 16 or 24 kHz and a bit rate from 6 to 40 Kbit/s. Main features:
• Compatibility with Skype for Business
• Flexible bitrate
• High quality
• Variety of sampling frequency
• Inband FEC and good resilience to Packet Loss
The phone’s TRANSFER hard key now by default performs Blind Transfer instead of Consultative Transfer.
A new logging option SIPE (a third-party Pidgin plugin for Microsoft Skype for Business client) was added to the System Logging page. This logging level may help with the investigation of cases related to Exchange integration.
Forcing PIN code authentication. Using a new configuration file parameter, the network administrator can force PIN code authentication; the only sign-in option will then be with user
extension number and PIN code. Allowing only the basic PIN code option on the user's phone helps avoid user mistakes and helps avoid storing the user password on the phone.
The SIP User Agent field was modified in compliance with Microsoft’s UC requirements. The modified information now appears in all SIP messages that have a User Agent ‘Header’ field. Until now, the User-Agent was (for example):
• AUDC-IPPhone-405_UC_2.0.13.205.15/1.0.0000.0
The new User-Agent is (for example):
• AUDC/3.0.0.575.42 AUDC-IPPhone-405_UC_3.0.0.575.42
Ability to make new calls during incoming calls. [Applicable to all phones except the 405HD and 420HD].
• The procedure of making a new outgoing call is not interrupted by an incoming call. The incoming call does not disrupt the user's number dialing process. The user can finish entering the digits and make their call, uninterrupted by the incoming call.
• Ability to ignore an incoming call by initiating a new one. When the user's phone rings indicating an incoming call, they can initiate a new call, ignoring the incoming, by pressing a
New Call softkey.
Liste des nouveautés, fonctionnalités supportés, ajustements, corrections dans la release notes LTRT-08293 400HD IP Phone Series for Microsoft Skype for Business Release Notes Ver. 3.0.0.575.42.pdf
Téléchargement ici: http://www.audiocodes.com/downloads/solutions
Microsoft a publié une nouvelle cumulative update pour ce mois de Mai 2017.
KB:3061064
Source: https://support.microsoft.com/en-us/help/3061064/updates-for-skype-for-business-server-2015
Nouvelles fonctionnalités:
Corrections:
Composants:
Intéressant:
The update supports administrators to move a meeting room object from a Microsoft Skype for Business Server 2015 on-premises environment to Microsoft Skype for Business Online in Microsoft Office 365 by running the Move-CsMeetingRoom cmdlet:
Téléchargement: https://www.microsoft.com/en-us/download/details.aspx?id=47690
Cela fait un moment que le modèle Audiocodes HD405 est en cours de certification pour Skype Entreprise, il semble que le processus touche a sa fin.
Le téléphone 405 SIP IP est un téléphone IP d’entrée de gamme, à bas prix, il offre les fonctionnalités essentielles permettant de répondre aux exigences actuelles, disponible en Gbe
- écran LCD, rétroéclairé, multilingue (132 X 64)
- 4 touches programmables
- AudioCodes Auto-provisioning
- Prise en charge complète du protocole SIP avec une large interopérabilité
- mécanismes de sécurité robustes
- Power over Ethernet (PoE)
- Prise en charge multilingue
- Surveillance de la qualité de la voix intégrée
- Connectivité casque et haut-parleur en « Full duplex »
Le firmware Skype Entreprise pour ce modèle est disponible au téléchargement en version 3.0.0.575.42:
La release notes devrait être en ligne d'ici la fin de semaine normalement.
Polycom a publier pour le mois de Mai 2017 une nouvelle révision pour le software Polycom UC en version 5.5.2.
La révision pour Skype Entreprise s'applique aux téléphones suivants:
Polycom® VVX® 201 business media phones
Polycom® VVX® 300/301/310/311 business media phones
Polycom® VVX® 400/401/410/411 business media phones
Polycom® VVX® 500/501 business media phones
Polycom® VVX® 600/601 business media phones
Polycom® SoundStructure® VoIP Interface
La révision comporte les nouvelles fonctionnalités suivantes:
● Enterprise Directory Default Search
● Registration Line Address in Status Bar
● BroadWorks Anywhere EFK for Soft Keys
● Hide Contact Directory and Favorites
● Personal Directory
● BroadSoft Server-Based Call Logs
● New Call Forwarding Icons
● Updated Do Not Disturb Icon
● Expanded Support for USB Headsets
● Support Added for CDP in VVX D60 Base Station
● ALLOW Header in 18x Provisional Responses
● Skype for Business SIP and Tel URI
Skype for Business SIP and Tel URI:
In Skype for Business environments, the phone places the last dialed method for a contact either through SIP URI or Tel URI and dials via the same method the next time until it reboots to the default SIP URI.
Web Sign In Using Skype for Business (disponible depuis 5.5.1)
You can enable web sign in on the phone using the parameter feature.webSignIn.enabled. After the parameter is configured, you can enable or disable web sign-in for your Skype for Business profile.
Device Lock Parameters
You can configure your phone using the up.configureDeviceLockAuthList parameter to set the order of display for Authorized/Emergency numbers, when device is locked in Skype for Business profile
Server Logging Levels for Skype for Business Server (disponible depuis 5.5.1)
In UC Software 5.5.2 and later, you can set the log levels for Polycom phones in a Skype for Business environment on the Skype for Business Server.
To set the server-side logging levels:
1 In the command shell, enter the command Set-CsUCPhoneConfiguration.
2 Set one of the following log levels.
- Off
- Low
- Medium
- High
Historique des dernières révisions:
La release note est disponible ici:
Applicatif:
CAB 5.5.2: http://support.polycom.com/content/support/North_America/USA/en/eula/ucs/uc-agreement-5-5-2-cab.html
5.5.2 Combined: http://support.polycom.com/content/support/North_America/USA/en/eula/ucs/uc-agreement-5-5-2-combined.html
5.5.2 Split: http://support.polycom.com/content/support/North_America/USA/en/eula/ucs/uc-agreement-5-5-2-split.html
Btoe 3.5.0: http://support.polycom.com/content/support/North_America/USA/en/eula/ucs/uc-agreement-btoe-3-5-0.html
La dernière version logiciel Skype Entreprise disponible au téléchargement pour les téléphones IP Audiocodes HD400 est la 2.0.13.205.30.2.
Aucune nouveauté en terme de fonctionnalité utilisateur.
Problèmes corrigés dans cette version.
- On rare occasions and only in specific environments, the user cannot hear the remote side at the beginning of a call for several seconds.
- Blind Transfer scenarios on rare occasions lead to dropped calls and phone reinitialization.
- A delay is noticed when answering incoming calls to a Response Group line.
- The phone configures its internal switch with VLAN tagged =1 when the external switch port is configured untagged (native) Vlan =1.
- Priority of ICE UDP host candidates in the SIP SDP packet is lower than TCP, causing the call to not be established.
- BToE:
• Users with a very specific user length (X*31 +1) cannot be paired.
• Escalation of a call from audio to video with a paired user may cause the call to disconnect.
- When the first incoming media packet is a Comfort Noise packet, voice on the incoming path may be inaudible because the configurable parameter 'prevent_CN_in_early_media' is configured to 0.
- Although log files are supposed to be cyclic and 50 KB in size, some may exceed their size limitation and cause flawed phone performance.
- On rare occasions, softkeys disappear after 24 days.
- The phone cannot redial a number that has a non-DID extension.
- In some environments, calls from PSTN occasionally disconnect immediately due to a mismatch in SRTP format.
- Some Hungarian characters cannot be displayed.
- Some Japanese characters cannot be displayed.
Plus d'informations ici: http://www.audiocodes.com/downloads
Toujours impatient de la future release officielle de la version 3.0 qui va amener beaucoup de nouvelle fonctionnalité.
Dans plusieurs projets de migration téléphonique Cisco vers Skype Entreprise, je dois conserver une partie de l'inventaire matériel existant comme des ATA.
La procédure suivante décrit le processus de reset factory d'un ATA Cisco 186.
Préalable;
Étapes:
1/ Décrocher le téléphone analogique
2/ Presser le bouton de configuration sur le boitier ATA
3/ Lorsque l'interface vocale pour la configuration déclenche, composer le code 322873738#* et * pour sauver
4/ Raccrocher le téléphone.
Attendre que la lumière termine de clignoter.
Dans plusieurs projets de migration téléphonique Cisco vers Skype Entreprise, je dois conserver une partie de l'inventaire matériel existant comme des ATA ou autres.
La procédure suivante décrit le processus manuel de migration d'un firmware SCCP vers SIP pour un Cisco ATA 186
Préalable;
Firmware SCCP existant:
Étapes:
1/ Dans le fichier zip correspondant au firmware SIP acquis, vous disposez de 3 programmes, en fonction de votre poste client, vous permettant de migration le firwmare du ATA vers SIP:
Le fichier stat186us.txt contient aussi des explications sur le processus.
Le firmware est sous format .zup
2/ Commande pour la mise a niveau
La commande liée au programme sata186us.exe permet les options suivantes:
usage: sata186us {-h[host_ip]} {-p[port]} {-quiet} <imageFile>
-h[host_ip] Set host IP to specific IP (in the case where there
are more than one IP addresses for the host.
Default use 1st IP address obtained by gethostbyname).
-p[port] Set server port to specific port (default is 8000,
use different port only if you are setting up an IP
directed upgrade server other than the default).
-quiet quiet mode, send all output to log file named
as [port].log (useful when running the upgrade
server as a deamon).
-any Allow upgrade even if software version is less
than or equal to those of client box.
-any2 Allow upgrade regardless of software type and version.
-d1,-d2,-d3 Set verbose level for debugging.
imageFile Image file is file with a '.zup' or '.kup' extension.
e.g.
sata186us -any -d1 test.zup
sata186us -h192.168.2.170 -p8002 -quiet test.zup
Exemple de la commande pour une mise a niveau, 10.8.1.100 étant mon poste de travail:
sata186us -any -h10.8.1.100 ATA030201SIP050616A.zup
Une fois le code obtenu, décrocher le téléphone analogique, appuyer sur le bouton suivant:
Composer le code.
Attendre que la lumière termine de clignoter, rester en ligne, une annonce vocale de fin de la mise a jour sera annoncé.
Votre Cisco ATA 186 est maintenant en version SIP:
Une nouvelle version de Attendant Pro pour Skype Entreprise est disponible au téléchargement: https://landiscomputer.freshdesk.com/support/solutions/articles/6000060064-attendant-pro-update-downloads
Versions:
Nouveautés:
Informations ici: https://attendantpro.blogspot.ca/2017/05/attendant-pro-q2-2017-update-transfer.html
Microsoft vient de publier une nouvelle mise à jour pour le client Skype for Business 2015 (Lync 2013).
KB:3191876
Lien: https://support.microsoft.com/en-us/kb/3141468
Date: 02/05/2017
Numéro de version: 15.0.4927.1000
Téléchargement:
Download the 64-bit Lync update package now
Prérequis:
Avant de procéder, vous devez installer cette mise a jour: Microsoft Office 2013 Service Pack 1 (KB2817430).
Corrections:
Depuis le 11 avril 2017, Exchange Server 2007 est en fin de vie.
Le produit a 10 ans maintenant, le temps passe vite, je me souviens encore mes premières intégrations UM avec du Cisco lors de la disponibilité de la technologie, pas de blog, documentation tres mince, support MS pas vraiment familier avec la téléphonie, mais ce fut formateur.
Que signifie un produit en fin de vie avec Microsoft:
Since the April 11, 2017, Microsoft will no longer provide:
Technical support for problems that may occur;
Bug fixes for issues that are discovered and that may impact the stability and usability of the server;
Security fixes for vulnerabilities that are discovered and that may make the server vulnerable to security breaches;
Time zone updates.
Informations ici: https://support.office.com/en-us/article/Exchange-2007-End-of-Life-Roadmap-c3024358-326b-404e-9fe6-b618e54d977d
Depuis la version Skype Cloud Connector Edition 1.4.2, vous pouvez maintenant assigner un certificat externe au serveur de Mediation. Cela vous permet ainsi d'établir une session TLS avec des passerelles de téléphonies.
Le certificat doit contenir la clé privée.
Set-ccexternalCertificateFilePath -Path "C:\SkypeCCE\Certificat\Edge.pfx" -Target EdgeServer
Set-CcExternalCertificateFilePath -Path "C:\SkypeCCE\Certificat\Mediation.pfx" -Target MediationServer
get-ccexternalCertificateFilePath
Microsoft a publié des mises a jour pour les modèles de Microsoft Lync Phone Edition suivant:
Polycom CX500, Polycom CX600 et Polycom CX3000:
Téléchargement: http://go.microsoft.com/fwlink/?linkid=203406
Aastra 6721ip et Aastra 6725ip
Téléchargement: http://go.microsoft.com/fwlink/?linkid=203407
HP 4110 et HP 4120:
Téléchargement: http://www.microsoft.com/en-us/download/details.aspx?id=28158
Corrections (Seulement les Polycom): KB 3194836 LCD display issues in Lync Phone Edition for Polycom CX500, Polycom CX600, and Polycom CX3000
Nouvelles fonctionnalités: This cumulative update fixes the issue in which 2017 DST changes for Chile and Turkey in Lync Phone Edition.
Version image: 7577.4531
