Set up Siemens C610 IP with VoiceOne

03 ott 2011

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C610 IP is the latest IP cordless phone model introduced by Siemens. It is able to handle up to three calls simultaneously, supports up to 6 handsets DECT GAP and allows you to configure up to 6 VoIP accounts. So let’s see in detail how to configur it with a VoiceOne system.

First you must create the extension in VoiceOne, for this example we added an extension in this way:

  • Name: Cordless
  • Number: 200
  • Password: 1234
  • VoiceOne IP: 192.168.1.100

 

Let’s go then to configure our cordless through its web interface. We enter in the Settings -> Telephony -> Connections and go to edit the first accounts available as follows:

In this way, after you save your changes by pressing the Set button, we should see the cordless correctly registered in VoiceOne and, from the phone’s web page we should see the following:

At this point we are going to set some advanced parameters that are used codecs:

and other parameters associated with the DTMF signaling and call transfer:

Now our Siemens C610 IP is configured and ready to use.

How to setup Patton SmartNode on VoiceOne

13 set 2011

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Patton-Inalp VoIP gateways allow you to integrate the traditional public network (analogue/ISDN/PRI) in the IP telephony business network. Thanks to these devices you can send and receive calls through traditional phone lines.

VoiceOne manages these devices like any other SIP VoIP providers, with the difference that is expected to be the device to register and not vice versa. For this reason we set up the VoiceOne Provider in Dynamic mode.

The Patton can be configured in two main ways:

  1. A SIP account for each physical line connected
  2. A SIP account for all lines connected in Hunt-Group

Here are the templates of optimal configuration for connection with VoiceOne, of the most popular models. For each model, except those offer the possibility to connect a single physical line, there are two templates, one for interfacing to VoiceOne as 1 SIP account – 1 physical line, and one for the interface mode 1 SIP account – all the physical lines in Hunt-Group.

 ISDN MODELS

 SN4552  SN4554 SN4634 SN4638
 NO Hunt-Group Patton-4552-R5.X-1Bri_1Account.txt Patton-4554-R5.X-2Bri_2Accounts.txt Patton-4634-R5.X-2Bri_2Accounts.txt  Patton-4638-R5.X-4Bri_4Accounts.txt
 Hunt-Group  Patton-4554-R5.X-2BriHG_1Account.txt Patton-4634-R5.X-2BriHG_1Account.txt Patton-4638-R5.X-4BriHG_1Account.txt

Within the templates will need to declare some variables dependent on your network configuration:

  1. VOICEONE_IP: VoiceOne ip address
  2. PATTON_IP: Patton device ip address
  3. PATTON_NETMASK: Patton device netmask
  4. GATEWAY_IP: gateway ip address of your network

SIP accounts are configured as follows:
USERNAME
: “isdn” in the case of single account, “BRI1, BRI2, etc. ..” in the case of multiple accounts
PASSWORD
: “isdn” to all accounts

 ANALOGUE MODELS

 SN4112  SN4114
 NO Hunt-Group Patton-4112-R5.X-2fxo_2Accounts.txt Patton-4114-R5.X-4fxo_4Accounts.txt
 Hunt-Group Patton-4112-R5.X-2fxoHG_1Account.txt Patton-4114-R5.X-4fxoHG_1Account.txt

Within the templates will need to declare some variables dependent on your network configuration:

  1. VOICEONE_IP: VoiceOne ip address
  2. PATTON_IP: Patton device ip address
  3. PATTON_NETMASK: Patton device netmask
  4. GATEWAY_IP: gateway ip address of your network

SIP accounts are configured as follows:
USERNAME
: “fxo” in the case of single account, “fxo1, fxo2, etc. ..” in the case of multiple accounts
PASSWORD
: “1234″ to all accounts

Voip PBX with Google Talk

08 set 2011

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The latest news in VoiceOne’s home is the cool integration with the Google’s world from Google Talk, the instant messaging text service, to Google Voice, its latest expansion in the telephony service. The version 1.8.420 opens the door to a world of possible integration between Google’s world and VoiceOne, some already implemented, others in the pipeline, and many still to think of.

With the last VoiceOne’s update has been integrated the configuration of Jabber and GTalk modules of Asterisk. The main improvements are reflected in the ability to create trunks and extensions with GTalk technology.

A GTalk trunk is just a phone line associated with an account of Google  platform and then a normal Gmail account or, if you use Google Apps, an email address of your company. This trunk can be used in input and output to call and be called from other GTalk accounts and, with the advent of Google Voice, to call outbound landlines and mobile numbers charging credit as it does with Skype. For outgoing calls the trunk behaves exactly as a traditional VoIP trunk while, for incoming traffic, it’s used to receive incoming calls from other GTalk users who call the account associated with the trunk.

A GTalk extension, however, is an extension number of the pbx that is associated with a GTalk account. Calling the extension number, the call will be routed by the system to GTalk user who will receive it through Gmail/Google Apps web panel, or on a client software that supports this type of technology. If a GTalk account configured as extension makes a call to a GTalk account associated with a trunk of the PBX, the system converts the email address of the caller extension number, allowing users to recall it in a totally transparent way.

And this is only the beginning. In fact it will soon be possible to have geographic numbers to be used through Google Voice, while it is already possible to chat with GTalk users through Asterisk macros created specifically to send and receive messages and text alerts. The gateway to the world of Google Apps is open!

VoiceOne 1.8.420 released

08 set 2011

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VoiceOne 1.8.420 has been released. View the changelog and download the new version.

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VoiceOne Localisation Portal

01 set 2011

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VoiceOne staff is pleased to announce the publication of the Web portal for the translation of VoiceOne. Reachable at http://pootle.voiceone.it, the portal is born to give the possibility, to open source community, to help realizing the full or partial translation of the VoiceOne’s Web GUI labels. So far VoiceOne has English translation (default) and Italian. Help us to add more languages ​​as possible in order to increase the spread of VoiceOne around the world.

How to setup Counterpath BRIA and X-Lite on VoiceOne

09 ago 2011

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One of the main advantages of VoIP PBX is the ability to use a software as a phone. A professional example of softphone is CounterPath Bria and its limited free version called X-Lite. In this tutorial we will see how to configure the softphone to register on VoiceOne, consult voice mailbox and LDAP directory (LDAP directory is not available on X-Lite but only on  Bria).

To register our softphone on VoiceOne we have to setup a new account on the application, corresponding to an extension previously created in VoiceOne. We’ll enter:

  1. Account Name: a fancy name for the account
  2. User ID: number of the configured extension in VoiceOne
  3. Domain: IP address of our VoiceOne
  4. Password: extension’s password

In the “Voicemail” section we can set up the access and monitoring of voice mail. To do this tick the “Check for Voicemail” option and insert the check number *98.

Save and enable the account. The softphone will register on VoiceOne allowing make and receive calls.

To configure LDAP access to centralized directory of our VoiceOne you must have activated the addon voiceone-ldap. Once done simply access the web interface of VoiceOne, in the configuration section under Extensions > LDAP. In this section you can get for each user its relative access username (dn); the password is the same as the extension, even before specified for SIP registration.

Obtained this value we start the Bria and choose Preferences > Directory. In the first screen “Server Settings” insert:

  1. Server: the IP address of our VoiceOne (eg. 192.168.1.50)
  2. Authentication method: Simple
  3. User name (dn=): the user identifier obtained from the LDAP page in  VoiceOne
  4. Root DN: dc=voiceone,dc=it
  5. Search expression: (&(cn=*)(hasSubordinates=FALSE))

In the second screen “Search Options” we’ll specify the “Search on Demand” to launch the application by pressing the search button.

In the third screen “Attribute Mapping” we’ll map the fields as specified in the figure below

Our softphone is now configured and ready to use. In the directory section we’ll find all contacts to do more research on them with.

How to setup Portech GSMBox MV-37X on VoiceOne

04 ago 2011

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In this article we’ll see how to configure a Gsmbox Portech MV-37X series with VoiceOne. These devices are characterized by an excellent quality / price ratio and allow you to interface GSM and UMTS lines to our PBX. Depending on the model it can include 1, 2, 4 or 8 SIM cards configurable on the PBX to be handled individually for incoming and outgoing calls. In this article we’ll see the configuration of an Portech MV-372 GSM (2 SIM), but the interface and the configuration mode is the same for the other models.

First we create a new VoiceOne VoIP Providers to allow registration of Gsmbox on the PBX. To do this, go to the Lines section, Providers and choose New Providers. Then enter a name for the new provider, select  Dynamic host type (it will be the Gsmbox to register on the PBX) and the host address empty, we value the field port with the range 5060-5062 (we will see later why) and left as SIP technology. Save.

Now we have to create two accounts, one for each SIM, that will allow us to call selecting which SIM to use and figure out which SIM the incoming calls  came from. To do this we click Add a new account link under the provider just created. We value, for simplicity, the fields name, username, password and confirm with sim1, select port 5060 and insert in Phone numbers the number 10 (which will be useful in Gsmbox for call routing). It is not necessary to specify any other data, save the page.

We create another account in the same way, for the second SIM. We fill up the fields name, username, password and confirm with sim2 and select the port 5062. Insert in Phone numbers the number 20 (which will be useful in Gsmbox for call routing). Save the page.
Now the configuration of the Gsmbox. Access the web interface using username and password authentication (voip/1234 are the defaults).

In the first section Route we define the incoming routes as below. In position 0 we’ll set as CID the value * and as URL the value 10 for the first SIM. In position 1, as CID we always indicate the value *, and as URL we set the value 20 for the second SIM.

We continue by defining the outgoing routes, this time the same for both SIM cards. Insert in the position 0 and 1 the URL * and the CallNum# as below.

We continue in the Mobile section. For both SIM cards select for SIP From field the value “Tel/Tel (No Reg)” and set the field CLID Presentation to ON, if we want to show our SIM number, or to OFF, if you want to hide it.

Let’s move into the SIP section, to interfacing with VoiceOne. In Service domain select first Mobile 1 and insert the parameters of the SIM1 account (created earlier in VoiceOne) accompanied by the IP address of our PBX. Then select Mobile 2 and insert in the same way the parameters of the account SIM2.

We check now that the ports associated with two accounts are the same as specified in VoiceOne (it should be by default). We should 5060 for Mobile 1 and 5062 for Mobile 2.

The last configuration parameter to set is in the SIP Responses. Here we set the “Response on Busy” to “503 Service Unavailable” and the “183 Session Progress” to ON that will allow us to hear ringing outgoing calls only from the exact moment when the phone starts to ring.

Pay attention to have successfully saved each screen and then save the configuration with the appropriate option.

The Gsmbox will restart by registering on our asterisk (VoiceOne) and making available for calls the two new trunks. Calls to and from the first SIM card will be routed on the trunk SIP/sim1, while those to and from the second SIM card will be routed on the trunk SIP/sim2.

VoiceOne 1.8.418 released

29 lug 2011

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VoiceOne 1.8.418 has been released. View the changelog and download the new version.

Fax Detect on VoiceOne

28 lug 2011

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The main change in version 1.8.417 VoiceOne, contained in all subsequent ones, is the addition of the Fax Detect feature. We can choose, indeed, if the pbx has to recognize fax calls, which are then differently handled by the system respect voice calls. The recognition on the call is rapid (almost instantaneous) and completely transparent to the caller that, in fact, does not realize that the system is checking the call.

First, you must enable Asterisk to make the detect of fax calls on SIP channels. To have this feature activated you must ensure that in VoiceOne under the section Technologies –> General Technology Options of the SIP technology, the field Faxdetect is checked and valued to yes (this setting  allows detecting both CNG and T38 faxes).

The Fax detect feature is activable in VoiceOne per line so, for every provider’s account you can indicate whether or not you want to have the fax detection for incoming calls. Within the account’s tab it will simply necessary tick the “Fax detect” contained in the “Settings” block and then select the incoming “Rule Set” where you want to redirect calls. The choice of the “Rule Set” allows us to decide which set of rules manage the fax call. As it was previously, calls received on the line are then processed with the “Rule Set” specified in the block of “Technology” and redirected to the “Rule Set” specified for fax calls if they are detected as a fax.

Let’s see step by step how to activate the fax detect on our line. Suppose in this example, you have an ISDN line, with only a public number, on which we receive both voice and fax calls. We choose this case only because it is  the most common, it does not matter if the line is ISDN, analog, VoIP or any other. To connect this line to VoiceOne we will surely have been created a provider for interconnection to a SIP Gateway (Patton, Autdiocodes, Grandstream, etc …) and an account in which we have specified the username and password registration, the numbers associated to the line and all what is need in the configuration. The novelty is the presence of the field Fax Detect. Tick ​​the field and select from drop down menu an incoming  Rule Set already created (in the screenshot we had previously created and left blank the Rule SetFaxDetected“). Save.

Now all calls we’ll receive on that line will be tested by the system to see whether they are fax or voice calls. If detected as a fax, incoming calls from the target line will follow the rules of the Rule Set, which we have just selected, that is “FaxDetected“. Let’s go then under “Rules” –> “Incoming Rules” to create a new rule that forwards all incoming calls that come in this rule set to extension 910, which is our fax extension that will transform the fax into a pdf and send it by mail.

Well, in this way if we receive a fax call on our line, the system will automatically recognize and handle as stated. Now some details:

  1. Is the fax addon required to use the fax detect? No, in our example we have forwarded the call to a fax extension, but we could forward it to any other extension, possibly related to an ATA device with an attached fax machine.
  2. In which time of the call is made the fax detect? Fax detection is made ​​after the macro that handles the call executes the command “Answer” answering the call. Only once it is answered, the system tests the channel and if it detects the presence of an incoming fax call diverts it to the Rule Set specified. Until then the flow of the call follows the rules set associated to the line in the “Technology” section. To let the system the time to detect the fax, simply insert within the macro, before the Dial command, these lines:
    exten = s,n,Answer
    exten = s,n,Wait(2)
  3. How is detected the fax? Responsible for this is the feature that is included natively in Asterisk 1.8.

Make good use

VoiceOne 1.8.417 released

27 lug 2011

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VoiceOne 1.8.417 has been released. View the changelog and download the new version.