CallViaPhone Advanced new feature for Outlook 2010/2013

Hi all,
I am pleased to announce a new version of CallViaPhoneAdvanced 2013 . As already mentioned in my previous posts,  CallViaPhone application,  through Lync interface, can be used to make and handle calls (similar to Better Together features of snom and polycom) to your phone (make call, close, hold, transfer, conference) , for more details please refer to this post CallViaPhone post detailed description .
Here below you can find new features of this version more in deep :

  • One of the most important new feature added is  that, from now, you can also use Microsoft Outlook 2010/2013, regardless of Lync. So the “call by phone” can be made ​​as described below:
    • From contacts in mail header or in appointment header:

mail-item

  • From Outlook Contacts Folder :

contact-item

  • New menu in configurator to use CallViaPhone with these ip-phone models :

PhoneModelsChoose

    • Snom Phones (default selection)
    • Cisco ip-phone (require some configuration on Cisco Call Manager, with this version we don’t have yet features like Close call, Hold Call, transfer call and conference call and Incoming call feature)
    • Yealink ip-phones (not available Incoming call feature)
    • Automatic check for new applications updates

You can download full version  (15 days limited) :

  • If you don’t have Lync  installed you can install any of these above for using Outlook callviaphone feature.
  • If you already have an old version of CallViaPhone , you don’t have to uninstall it, just install over.

For any questions don’t hesitate to contact sales@callviaphoneapp.com or support@callviaphoneapp.com

Call Via Phone ADVANCED for Lync 2010 & Lync 2013 , deep review and download link

Hi all ,

As i wrote in my previous article that you can find here : CallViaPhoneAdvanced  , this plugin is developed for SNOM phone and Lync 2010/2013 and permit to control the phone with right click on a contact in Lync (Lync contacts, Outlook contacts , Active Directory contacts) and start calls directly from the phone, like CTI (Computer Telephony Integration) .

We know that in WPC 2013 , SNOM demonstrate Enhanced Better Together that could be available at the end of 2013 , in Matt Landis blogpost here , you can find a detailed description.  This new SNOM feature add a new driver in Lync that let you choose what default Audio device use for every call that you made from lync , these calls start from the phone , also when you receive a call you pick up the call from the phones and not from the client (if you have already configured  the snom audio device in Lync down- left menu ), but a good things is that you when you are in conversation , you can switch this call between Lync client and snom phone.

So what’s the difference between CallViaPhoneAdvanced and SNOM EBT ? There are more than one difference and benefits :

  • CallViaPhone Advanced doesn’t not require Lync user to have Enterprise voice for Outgoing call functionality (for example if you use snom phone connected to other vendors like asterisk, cisco ,etc.. and you use Lync with or without PBX integration )
  • When you start a call , if you want to decide call per call which phone use (snom phone or lync client) with CallViaPhoneAdvanced you can.
  • CallViaPhone Advanced allows  future customization and it’s in continuous developing  (Programming Phone keys, full interaction with any functionality available on the Phone)

Thanks to Michael LaMontagne,  a very skilled Lync Expert in Canada , i also develop the incoming call feature that let you to answer an inbound call from the phone, this simple adding another toast in left side of Lync toast with an “Answer by Phone” button, simply and immediate for user experience. But starting from this imagine how many customization we can implement :

  • when and incoming call arrive, start an external application with caller information (CRM?)
  • Busy on second call or based on presence status

In addition i want to encourage who want try the application (15 days limited) to download from link below  :

For last version please refer to this post.

if you want to learn more you can find an awesome review here made by  Michael LaMontagne.

Don’t hesitate to comment or ask information about it .

PBX replacement with MS Lync (with Dual Forking) Part 2

As i mention in my last post (part 1) we can choose to use the Voice gateway in pass-through as shown here :

PBXReplacementPart2_Example_base scenario

but to do this , we have to consider various steps before moving the IP-PBX in production  and insert voice gateway between PSTN and the enviroment, so let start from the beginning,  these are necessary steps :

1.  Voice gateway configuration for PSTN trunk (only configuration not yet connected), in the other side (to lync and to PBX) configuration of SIP trunk to Lync and SIP trunk to IP-PBX (IMPORTANT : to obtain a dual forking of calls in this scenario we must use only sip trunk to and from  PBX , so we must have an IP-PBX with sip trunk enabled ).

At this step we don’t have any disservice on IP-PBX production environment but we are ready to switch PSTN from IP-PBX to Voice gateway .PBXReplacement_part2_Example_step1

2.  We can schedule session tests to verify that all configuration made before work    fine (for example during time range in which we couldn’t have any outages to users, for example during the night?), in this way if not all scheduled tests list will be fine , we can rollback easily and move PSTN trunk to IP-PBX again.

To achieve this we must locate the Voice gateway properly sized, to mantain compliance on  requirement in terms of business continuity, for example redundant power supply , right number of SIP channels to/from lync, and to/from IP-PBX.

In this way we can also test a SIP trunking from a provider instead of PSTN (or buy another PSTN trunk to do a test pilot for Lync voice for example) , because until we switch the PSTN from IP-PBX to Voice Gateway, we can work easily in Voice Gateways side without give any problem in IP-PBX side.

About call flow management and dual forking we have the same behavior as i wrote in my previous post (part 1) , unique difference is that all configuration for dual forking is made in Voice gateways side , and in PBX side we have only to switch all inbound and outbound call to the new SIP trunk instead of PSTN trunk already switched off.

At this point we can assert that we have a lot of way to do a soft migration or simply to use the existence telephony infrastructure for Lync and generally Microsoft UC. We know that Microsoft Lync just for presence/audio/video/conference is really wasted and with a good approach and a small effort we can implement an excellent Lync Voice project for a really Unified Communications experience.

PBX replacement with MS Lync (with Dual Forking) Part 1

Talking about PBX replacement with MS Lync can be a difficult argument when proposed to customers. But as the nature of MS Lync we have a lot of ways to do it. Usually we can meet two different type of customers, one can think that his employees must change how they work day by day, and for this reasons we can explore solution with direct switch to new technology providing a direct cut-off ; the other one,  not so confident,  prefer a soft migration and possibly a true coexistence between old and new phone system, the last one obviosly is more complicated,  but surely the most funny for us:-), i want to explain  you how we can do a soft migration also with a good coexistence, for now i can mention 2 type of IP-PBX or TDM-PBX: ALCATEL OXE and Cisco CCM.

The first important thing is that all of this project must provide a Voice Media Gateway to ensure that all translation and, eventually transcoding,  from one system to Lync and viceversa don’t drive us crazy…:-)

Actual Infrastructure Enviroment without integration

PBXReplacementExample_base scenario

Based infrastructure consider that we have a fully up and running Lync enviroment and a consolidate Phone infrastracture .

Scenario  (Coexistence with Dual Forking)

If we have a ALCATEL OXE (with remote extension license), CISCO CCM (sip forking with extension mobility license) or a TDM/IP-PBX that support forking to another number not included in its dial-plan (for example to a sip trunk or TDM trunk connected) we can consider this scenario :

PBXReplacementExample_scenario1

Using  Voice Gateway between Lync and Phone infrastructure give us a lot of configuration that otherwise we could not easily do without provide a big effort from the Phone system team .

In this scenario we can consider this events :

Inbound call from PSTN : When we receive a call from PSTN to +3906….4444 , call arrive to PBX , PBX at this stage send the call to the extension in its dialplan and see that there’s also another number associated (for example 9994444) and , in parallel , fork this call to that number with 999 (a prefix trunk associated to the Voice gateway).

When the call arrive to Voice Gateway with destination number 9994444 , it translate called number in +3906….4444 and send to lync .

Result  :  Lync client (or lync phone) and PBX phone ring at the same time, and when one of this two pick up the call ,the other one stop ringing .

Inbound call from another PBX phone : When we receive a call from another PBX phone  to 4444 , call arrive to PBX , PBX at this stage send the call to the extension 4444 and see that there’s also another number associated (for example 9994444) and , in parallel , fork this call to that number with 999 (a prefix trunk associated to the Voice gateway).

When the call arrive to Voice Gateway with destination number 9994444 , it translate called number in +3906….4444 and send to lync .

Result  :  Lync client (or lync phone) and PBX phone ring at the same time, and when one of this two pick up the call ,the other one stop ringing .

Outbound call from Lync to other PBX phone : In lync we have two way to make a call to a contact, if we make a classic Lync call , this call remain inside Lync enviroment but if we make a work phone the call is translated for example in extension format :

– Digited : +3906……4444 , normalized in 4444  so the call are sent outside Lync through the Voice Gateway  and arrive to the extension 4444 , as i mention before in pbx enviroment 4444 have another extension configured (9994444) that corrisponds to the Voice gateway trunk and the same call was also diverted to Lync client .

Result  :  Lync client (or lync phone) and PBX phone ring at the same time, and when one of this two pick up the call ,the other one stop ringing .

Yes i know , a little cumbersome but it’s work fine .

Inbound call from lync to PBX and dual forked to lync

Outbound call from Lync to other PSTN: All external calls made from Lync follow the classic flow to PSTN (Voice Gateway –> PBX –> PSTN) , it’s important to know that all calls made by Lync can have the same Calling number as the associated extension in PBX dial plan :

– for example if i make a call from PBX phone my external DID is : +3906……4444, but PBX add instead of me the +3906….. (* maybe that +39 is not considered in a national call) .

When i make a call from lync if i want that it must be the same calling number as the PBX phone ,  i have to configure on Voice Gateway a good format for PBX to accept DID so for example in ALCATEL enviroment i must pass to it the call in this format :

calling number (Lync side) +3906…..4444  — >  Translated by VG in  : 06…..4444 , in this way ALCATEL recognize the call as one from its dial-plan, otherwise can appean that my calling number is only +3906……. without the extension.

Result : the call appear to PSTN exactly from one number shared by Lync and PBX and we can realize a true Single Number Reach

Requirement for this scenario

We have to consider that if we make a QSIG trunk between PBX and Voice Gateway my advice is to use a QSIG-GF (Generic Function) not basic because there are a lot of service such as call diversion, line identification, etc.. that is not implemented on QSIG-BC (Basic Call).

If we choose a SIP trunk between PBX and Voice Gateway we have to consider in Voice Gateway side licenses for IPtoIP and eventually transcoding with DSP onboard because if we configure trunk from/to PBX in a RTP codec different from G711, for example G729 , all calls are trascoded (Lync Mediation server work only in G711).

I hope that this post can be useful for you , and please don’t hesitate to comment 🙂

In part 2 we’ll consider a scenario in which I’ll describe the positioning of Voice Gateway in passthrough between PSTN and PBX to prepare a clean migration phase.