MiRTA PBX documentation - User contributions [en]

Provisioning

2016-06-24T17:56:24Z

Manager:

Provisioning of phones is phone independent, meaning MiRTA PBX has no knowledge of the various brands or models, but offers a generic framework to create a template and use some variables. Some variables are already stored in the system, but more can be added easily. Provisioning is made over HTTP and HTTPS. Obviously it is preferable to provision over HTTPS to avoid having SIP credentials to travel in clear over the Internet. Some brand of phones are quite picky about the certificate authority used for the SSL, so in case of problems, it can be good to make a test using HTTP or check the phone configuration about third party CA.

Some phone templates are provided as proof of concept, but they are not supposed to be neither complete neither accurate.

== Phone Models ==

Several phone models can be defined. For each phone model the number of lines needs to be defined, usually with the starting and ending line number. Please refer to the phone's provisioning guide to customize or create new templates.

[[file:phonemodes.png|thumb|400px]]

A phone template can be "locked" to prevent any accidental modification. You can easily duplicate a provisioning template using the action icon on the right

For each phone you can define a generic name, an option description, the first and last account number, the mac request template, the directory request template, the remote provisioning POST message along with the credentials to use for remote provisioning. Remote provisioning is the ability of some manufacturer to deliver a phone configuration just out of the box, like Yealink RPS and Polycom ZTP.

The template may contain variables in the general format ${name|default|type|options}.

Some variables are predefined and are read directly from the system configuration:

* line_active – Always 1
* displayname – Name of the Extension assigned
* virtualdisplayname – Name of the Virtual Extension assigned
* username – SIP username
* authname - SIP username
* number – Number of the Extension assigned
* virtualnumber – Number of the Virtual Extension assigned
* secret – Password of the Extension assigned
* index - The position of the account starting from the first line defined
* tenantcode - The account code for tenant
* subscribemwi - The option configured for "Send MWI only if subscribed"
* mailbox - The mailbox associated to the extension
* transport - The transport used for the extension (asterisk notation: udp, tcp and tls)
* transportY - The transport used for the extension (Yealink notation: 0 for udp, 1 for tcp and 2 for tls)
* transportP - The transport used for the extension (Polycom notation: UDPOnly for udp, TCPOnly for tcp and TLS for tls)
* codecs - The codecs configured for the extension (asterisk notation)
* line – Order position of the line (usually 1, 2, 3, etc)
* line_m1 - Order position of the line (usually 1, 2, 3, etc)
* key - Order position of the line (usually 1, 2, 3, etc)
* mac – Mac Address of the phone, without :
* provisionpassword – Password for provisioning URL

It is possible to define loops in the template using the {line_loop} and {/line_loop} keywords. Each variable used needs to be defined in the “Variables” menu. In case a variable is not
defined, an alert message will be shown. A default value can be defined for the variable, using the | character, like: {$server_host|demo.mirtapbx.com} To help users in configuring their system, you can specify the editing box type, so if by default is a text type as in:

Admin section

2016-06-24T17:54:33Z

Manager:

[[Providers]]

[[Billing]]

[[Provisioning]]

[[Global Configurations]]

[[PBX Nodes]]

[[Routing Profiles]]

[[CallerID Mods]]

[[Tenants]]

[[Users]]

[[User Profiles]]

[[DIDs List]]

[[Security]]

[[Admin Stats]]

[[Translations]]

[[Themes]]

[[Defaults]]

[[Settings]]

File:FlowBBKYealink.png

2016-06-22T20:14:49Z

Manager:

Select the destination for DID based on the BLF selected on the phone

2016-06-22T20:14:21Z

Manager:

The client want to manually select one of three optional destinations for his DID. Option 1 will be Voicemail 1000 (weekend), option 2 will be Voicemail 2000 (handmatig) and option 3 will be a weektime condition.

I start by creating three Flows, named 10000, 20000 and 30000 monitoring the relative extension.

After creating the three Flows, I create a series of Custom Destination "Set Extension in use" and "Set Extension not in use" to set each of the three flows in use and not in use

[[File: flowBBK.png|400px]]

[[File: flowBBK2.png|400px]]

[[File: flowBBK3.png|400px]]

When each flow is activated, it set his associated extension to IN USE and move all other to NOT IN USE

The Custom destinations are configured as following:

[[File: flowBBKsetinuse.png | 400px]]

Now it is time to define a condition for each of the flow, if the flow extension is IN USE, the destination will be followed, like as following:

[[File: flowBBKcondition.png | 400px]]

The DID will just list all the condition, one after the other

[[File: flowBBKDID.png | 400px ]]

On the phone, the BLF will be configured as following (Yealink)

[[File: flowBBKYealink.png | 400px]]

File:FlowBBKDID.png

2016-06-22T20:13:39Z

Manager:

Select the destination for DID based on the BLF selected on the phone

2016-06-22T20:12:45Z

Manager:

File:FlowBBKcondition.png

2016-06-22T20:11:52Z

Manager:

Select the destination for DID based on the BLF selected on the phone

2016-06-22T20:11:21Z

Manager:

File:FlowBBKsetinuse.png

2016-06-22T20:10:18Z

Manager:

Select the destination for DID based on the BLF selected on the phone

2016-06-22T20:09:42Z

Manager:

Select the destination for DID based on the BLF selected on the phone

2016-06-22T20:08:58Z

Manager:

File:FlowBBK3.png

2016-06-22T20:07:44Z

Manager: Manager uploaded a new version of File:FlowBBK3.png

File:FlowBBK3.png

2016-06-22T20:06:57Z

Manager:

File:FlowBBK2.png

2016-06-22T20:06:12Z

Manager:

Select the destination for DID based on the BLF selected on the phone

2016-06-22T20:06:03Z

Manager:

Select the destination for DID based on the BLF selected on the phone

2016-06-22T20:04:57Z

Manager:

File:FlowBBK.png

2016-06-22T20:04:31Z

Manager:

Select the destination for DID based on the BLF selected on the phone

2016-06-22T20:03:58Z

Manager: Created page with "The client want to manually select one of three optional destinations for his DID. Option 1 will be Voicemail 1000 (weekend), option 2 will be Voicemail 2000 (handmatig) and o..."

Configuration Guides

2016-06-22T19:58:56Z

Manager:

Here I am trying to collect some of your requests with the way used to solve your needs. Take in mind there can be multiple ways to do the same trick and mine cannot be the best one.

[[Read a number from the caller and run an AGI script using it as parameter]]

[[Change the CallerID of the extension when dialing out]]

[[Select the destination for DID based on the BLF selected on the phone]]

Changing the clock source

2016-06-19T16:12:13Z

Manager: Created page with "Selecting the best clock source can improve call quality, especially for conferences where the clock source is really important. You can check which clock source is used by y..."

Selecting the best clock source can improve call quality, especially for conferences where the clock source is really important.

You can check which clock source is used by your system:

cat /sys/devices/system/clocksource/clocksource0/current_clocksource

You can list the available clock sources for your system:

cat /sys/devices/system/clocksource/clocksource0/available_clocksource

You can change the clock source temporarly, until next reboot. Here as example, acpi_pm is selected

echo "acpi_pm" > /sys/devices/system/clocksource/clocksource0/current_clocksource

If you need to permanent change the clock source, you need to edit /etc/grub.conf

title CentOS (3.4.53-8.el6.centos.alt.x86_64)
root (hd0,1)
kernel /vmlinuz-3.4.53-8.el6.centos.alt.x86_64 ro root=/dev/md2 rd_NO_LUKS rd_NO_DM nomodeset crashkernel=auto SYSFONT=latarcyrheb-sun16 LANG=en_US.UTF-8 KEYTABLE=de clocksource=acpi_pm
initrd /initramfs-3.4.53-8.el6.centos.alt.x86_64.img

== An overview on hardware clock and system timer circuits ==

When it comes to talk about a system's clock, the hardware sits at the very bottom. Every typical system has several devices, usually implemented by clock chips, that provide timing features and can serve as clocks. So, which hardware is available depends on the particular architecture. The clock circuits are used both to keep track of the current time of the day and to make precise time measurements. The timer circuits are programmed by the kernel, so they issue interrupts at a fixed, and predefined, frequency. For instance, IA-32 and AMD64 systems have at least one programmable interrupt timer (PIT) as a classical timer circuit, which is usually implemented by an 8254 CMOS chip. Let's briefly describe the clock and timer circuits that are usually found with any nearly modern system of those architectures:

=== Real Time Clock (RTC) ===
The RTC is independent of the system's CPU and any other chips. As it is energized by a small battery, it continues to tick even when the system is switched off. The RTC is capable of issuing interrupts at frequencies ranging between 2 Hz and 8,192 Hz. Linux uses the RTC only to derive the time and date at boot time.

=== Programmable Interrupt Timer (PIT) ===
The PIT is a time-measuring device that can be compared to the alarm clock of a microwave oven: it makes the user aware that the cooking time interval has elapsed. Instead of ringing a bell, the PIT issues a special interrupt called timer interrupt, which notifies the kernel that one more time interval has elapsed. As the time goes by, the PIT goes on issuing interrupts forever at some fixed (architecture-specific) frequency established by the kernel.

=== Time Stamp Counter (TSC) ===
All 80x86 microprocessors include a CLK input pin, which receives the clock signal of an external oscillator. Starting with the Pentium, 80x86 microprocessors sport a counter that is increased at each clock signal, and is accessible through the TSC register which can be read by means of the rdtsc assembly instruction. When using this register the kernel has to take into consideration the frequency of the clock signal: if, for instance, the clock ticks at 1 GHz, the TSC is increased once every nanosecond. Linux may take advantage of this register to get much more accurate time measurements.

=== CPU Local Timer ===
The Local APIC (Advanced Programmable Interrupt Controller) present in recent 80x86 microprocessors provide yet another time measuring device, and it is a device, similar to the PIT, which can issue one-shot or periodic interrupts. There are, however, a few differences:

* The APIC's timer counter is 32 bit long, while the PIT's timer counter is 16 bit long;
* The local APIC timer sends interrupts only to its processor, while the PIT raises a global interrupt, which may be handled by any CPU in the system;
* The APIC's timer is based on the bus clock signal, and it can be programmed in such way to decrease the timer counter every 1, 2, 4, 8, 16, 32, 64, or 128 bus clock signals. Conversely, the PIT, which makes use of its own clock signals, can be programmed in a more flexible way.

=== High Precision Event Timer (HPET) ===
The HPET is a timer chip that in some future time is expected to completely replace the PIT. It provides a number of hardware timers that can be exploited by the kernel. Basically the chip includes up to eight 32 bit or 64 bit independent counters. Each counter is driven by its own clock signal, whose frequency must be at least 10 MHz; therefore the counter is increased at least once in 100 nanoseconds. Any counter is associated with at most 32 timers, each of which composed by a comparator and a match register. The HPET registers allow the kernel to read and write the values of the counters and of the match registers, to program one-shot interrupts, and to enable or disable periodic interrupts on the timers that support them.

=== ACPI Power Management Timer (ACPI PMT) ===
The ACPI PMT is another clock device included in almost all ACPI-based motherboards. Its clock signal has a fixed frequency of roughly 3.58 MHz. The device is a simple counter increased at each clock tick. However the ACPI PMT is preferable to the TSC if the operating system or the BIOS may dynamically lower the CPU's frequency or voltage. When this happens, TSC's frequency changes causing time warps and others side-effects, while the frequency of ACPI PMT does not.

Configuring Zoiper to use chat

2016-06-19T16:05:54Z

Manager:

'''Unfortunately latest Zoiper version uses some additional SIP message formats breaking the compatibility with Asterisk'''

Zoiper is a common and stable SIP client available for multiple platform, including Windows. It has a nice chat feature using SIP SIMPLE protocol, supported by Asterisk and MiRTA PBX. Using the SIP SIMPLE protocol you can exchange messages with almost all kind of desktop and soft phones, like they were SMS. Configuring Zoiper to use that requires few extra steps.

Start by configuring an account on Zoiper. All other options are default.

[[File:ZoiperAccount.png|400px]]

To be able to chat with an extension, this one needs to be configured in the contacts and available

[[File:ZoiperContact.png|400px]]

The extension needs to be configured for presence as following

[[File:ZoiperContactConfig.png|400px]]

Setup Guides

2016-06-19T16:04:16Z

Manager:

* [[Running on VMware]]
* [[Upgrading kernel]]
* [[Using Parking Lots]]
* [[Clustering and High Availability]]
* [[Configuring Yealink to receive the number from the parked call]]
* [[Configuring Zoiper to use chat]]
* [[Changing the clock source]]

File:Dialasgeneric.png

2016-06-18T19:27:54Z

Manager:

Change the CallerID of the extension when dialing out

2016-06-18T19:27:43Z

Manager: /* Use a feature code */

You may have the need to call out using different CallerID, some of them may be not even defined in the system. There are multiple ways to do it, let's review them:

== Use CallerID regex ==

If you have some "rules" for choosing the CallerID, you may use the CallerID regex located in the DIDs screen

[[File:calleridregextop.png|400px]]

and then associate the Outbound CallerID regex created to your extension. This way you can use an Italian CallerID number when dialing a number starting for 39 or use an American number when dialing a number starting for 1

== Use Forward To custom destination with a feature code ==

You can start by creating a custom destination "Forward To" setting the forward number to the special variable {$PARAM1} and picking up or editing the CallerID for each one of your custom CallerID

[[File:forwardtocustomcallerid.png|400px]]

Then you'll create a different feature code for each one of your custom destinations

[[File:featurecodeforzurichcallerid.png|400px]]

== Use a feature code ==

You can easily create a feature code for each of your custom CallerID. Start by creating a Custom Destination, type "Alter CallerID number" for each one of your custom CallerID

[[File:altercalleridzurichoffice.png|400px]]

then create a Dial As feature code to just dial out with the dialed number

[[File:dialasgeneric.png|400px]]

and then define a feature code to dial out by using them

[[File:dialasusingzurichoffice.png|400px]]

File:Dialasusingzurichoffice.png

2016-06-18T19:26:20Z

Manager:

Change the CallerID of the extension when dialing out

2016-06-18T19:26:08Z

Manager: /* Use a feature code */

You may have the need to call out using different CallerID, some of them may be not even defined in the system. There are multiple ways to do it, let's review them:

== Use CallerID regex ==

If you have some "rules" for choosing the CallerID, you may use the CallerID regex located in the DIDs screen

[[File:calleridregextop.png|400px]]

and then associate the Outbound CallerID regex created to your extension. This way you can use an Italian CallerID number when dialing a number starting for 39 or use an American number when dialing a number starting for 1

== Use Forward To custom destination with a feature code ==

You can start by creating a custom destination "Forward To" setting the forward number to the special variable {$PARAM1} and picking up or editing the CallerID for each one of your custom CallerID

[[File:forwardtocustomcallerid.png|400px]]

Then you'll create a different feature code for each one of your custom destinations

[[File:featurecodeforzurichcallerid.png|400px]]

== Use a feature code ==

You can easily create a feature code for each of your custom CallerID. Start by creating a Custom Destination, type "Alter CallerID number" for each one of your custom CallerID

[[File:altercalleridzurichoffice.png|400px]]

and then define a feature code to dial out by using that

[[File:dialasusingzurichoffice.png|400px]]

File:Featurecodeforzurichcallerid.png

2016-06-18T19:24:46Z

Manager:

Change the CallerID of the extension when dialing out

2016-06-18T19:24:34Z

Manager: /* Use Forward To custom destination with a feature code */

You may have the need to call out using different CallerID, some of them may be not even defined in the system. There are multiple ways to do it, let's review them:

== Use CallerID regex ==

If you have some "rules" for choosing the CallerID, you may use the CallerID regex located in the DIDs screen

[[File:calleridregextop.png|400px]]

and then associate the Outbound CallerID regex created to your extension. This way you can use an Italian CallerID number when dialing a number starting for 39 or use an American number when dialing a number starting for 1

== Use Forward To custom destination with a feature code ==

You can start by creating a custom destination "Forward To" setting the forward number to the special variable {$PARAM1} and picking up or editing the CallerID for each one of your custom CallerID

[[File:forwardtocustomcallerid.png|400px]]

Then you'll create a different feature code for each one of your custom destinations

[[File:featurecodeforzurichcallerid.png|400px]]

== Use a feature code ==

You can easily create a feature code for each of your custom CallerID. Start by creating a Custom Destination, type "Alter CallerID number" for each one of your custom CallerID

[[File:altercalleridzurichoffice.png|400px]]

and then define a feature code to dial out by using that

File:Forwardtocustomcallerid.png

2016-06-18T19:23:28Z

Manager: Manager uploaded a new version of File:Forwardtocustomcallerid.png

Change the CallerID of the extension when dialing out

2016-06-18T19:23:01Z

Manager:

You may have the need to call out using different CallerID, some of them may be not even defined in the system. There are multiple ways to do it, let's review them:

== Use CallerID regex ==

If you have some "rules" for choosing the CallerID, you may use the CallerID regex located in the DIDs screen

[[File:calleridregextop.png|400px]]

and then associate the Outbound CallerID regex created to your extension. This way you can use an Italian CallerID number when dialing a number starting for 39 or use an American number when dialing a number starting for 1

== Use Forward To custom destination with a feature code ==

You can start by creating a custom destination "Forward To" setting the forward number to the special variable {$PARAM1} and picking up or editing the CallerID for each one of your custom CallerID

[[File:forwardtocustomcallerid.png|400px]]

Then you'll create a different feature code for each one of your custom destinations

== Use a feature code ==

You can easily create a feature code for each of your custom CallerID. Start by creating a Custom Destination, type "Alter CallerID number" for each one of your custom CallerID

[[File:altercalleridzurichoffice.png|400px]]

and then define a feature code to dial out by using that

File:Forwardtocustomcallerid.png

2016-06-18T19:19:21Z

Manager:

Change the CallerID of the extension when dialing out

2016-06-18T19:19:10Z

Manager:

You may have the need to call out using different CallerID, some of them may be not even defined in the system. There are multiple ways to do it, let's review them:

== Use CallerID regex ==

If you have some "rules" for choosing the CallerID, you may use the CallerID regex located in the DIDs screen

[[File:calleridregextop.png|400px]]

and then associate the Outbound CallerID regex created to your extension. This way you can use an Italian CallerID number when dialing a number starting for 39 or use an American number when dialing a number starting for 1

== Use Forward To custom destination with a feature code ==

You can start by creating a custom destination "Forward To" setting the forward number to the special variable {$PARAM1} and picking up or editing the CallerID for each one of your custom CallerID

[[File:forwardtocustomcallerid.png|400px]]

== Use a feature code ==

You can easily create a feature code for each of your custom CallerID. Start by creating a Custom Destination, type "Alter CallerID number" for each one of your custom CallerID

[[File:altercalleridzurichoffice.png|400px]]

and then define a feature code to dial out by using that

File:Altercalleridzurichoffice.png

2016-06-18T19:05:37Z

Manager:

Change the CallerID of the extension when dialing out

2016-06-18T19:05:25Z

Manager:

DIDs

2016-06-18T18:57:59Z

Manager: /* Outbound CallerID Regex */

[[File:Did.png|thumb|400px]]
To every tenant can be assigned a group of DIDs, numbers to be dialed from the outside. 
Each DID can be configured to reach a certain extension or Queue or Hunt Group or IVR or any combination of the elements defined for the tenant. 
The same DID cannot be assigned to more than one tenant and there is no correlation between the dialing profile assigned to a tenant, the trunk provider used and the incoming DID. 
For this reason is highly important that the incoming INVITE from the trunk provider contains all the info to identify correctly the DID it is addressing to. 

Due to the fact there is a lack of strong standardization among trunk providers on how to transmit the DID information, a “best guess” is used to identify the correct DID, using any of the format currently in use around the world.

A DID can be configured to accept voice, fax or try to guess the calling party (if voice of fax). 
Autodetection relies on signalling at the start of the call and cannot be always reliable. 
The time to detect the fax tone can be set in Admin/Settings page.

== Editing New/Define DIDs ==

[[File:newDID.png|thumb|400px]]

The number must be entered in three parts, starting with international prefix, area code and number. Even if your telephone standard doesn't allow the usage of any “short” form for dialing the number, requesting for example, to dial always the area code, the number is requested to be entered this way. Using the Admin/Settings menu is possible to enable the use of a “single box DID” to just enter the number in a single box, but this is highly discouraged. 
It is possible to use regular expression inside the number. In case multiple regular expressions match the number dialed, the ordering is based alphabetically on the Comment. 
'''Comment''' is just a comment and it is not used in any way. 
'''Max channels''' allows you to set the maximal number of channels available on the DID. Any additional call received will be refused with a busy signal. 
'''Use CNAM service''' allows you to assign the Caller ID Name based on this popular service offered in most countries. You can configure the service in Admin/Settings page. 
'''Use as Emergency CallerID''' permits you to mark this DID as one of the available to be used when an emergency call is dialed. Due to the fact the emergency numbers are not standard among all countries, it is your duty to assign the “emergency flag” to the dialing rules for emergency numbers. When a call is dialed and it is using a dialing rule marked as “emergency”, the Caller ID number used for the extension is the one defined as “Emergency” among the ones with this flag set. 
'''Inbound Call Rate''' allows you to define a call rate for inbound calls.

== Voice ==

This section allows to define how to route the call when a voice call is received. 
'''Unconditional Forward''' is a destination that can be set on the DID to send the call to a particular destination. It is enable/disabled by the checkbox and that checkbox can be easily controlled by a feature code. This is not the destination to be set for common usage, you need to use the one in the Voice section. 
'''Always record''' permits to force the recording of the call, whichever is the future of the call. Recordings will be available through the Call History menu. 
'''Email recordings''' to defines if and to who the recordings needs to be sent once the call is over. Multiple destinations email can be entered with any delimiter, like space, comma and point and comma. 
'''Minimum Size (Bytes)''' allows to receive those recordings bigger than the size set, in Bytes. Recording takes place only on bridged channels, so IVR prompts or Music On Hold will be not recorded. 
Using '''the Prefix CallerID Num''' is possible to define a string to be added to any CallerID number received to identify for example the DID the call is coming from. The same result can be achieved by defining a Custom Destination. 
'''Prefix CallerID Name''' is the same as above, but for the Caller ID Name. 
'''Destination multiple drop down''' allow to define the list and the order of the objects receiving the call.

== Fax ==

This section allows to define what to do with the call if a fax is received on the number defined. Receiving Fax over the Internet has often a low success rate, even if T.38 protocol is used.

[[File:didfax.png|thumb|400px]]

'''Receive fax''' can be used to select if autodetect, force or disallow the reception of a Fax over the current DID.

'''Fax Station ID''' and '''Fax Header''' allows you to customize your virtual fax.

'''Fax Protocol''' can be selected among T.38 (the preferred way to receive faxes), T.38 with fallback to G.711 or only G.711.

'''Email destinations box''' permits to enter one or multiple comma delimited email address to forward the received fax. Received fax are sent in PDF format. Partially received faxes are sent in the same way.

It is possible to store the fax received for later reviewing using the '''Store Fax Received control'''.

== SMS ==

[[File:sms.png|thumb|400px]]

You can specify how to pull or receive SMS on this DID.

'''Protocol''' lets you choose the protocol to be used. More info can be found in embedded help.

The SMS can be then delivered to a SIP phone as SIP message or to any other destination.

== Bulk DIDs creation ==

[[File:Bulkdid.png|thumb|400px]]

It is possible to create multiple DIDs at once with Bulk DIDs creation button.

In this way a range of number is requested.

All the generated numbers are configured in the same way.

== Use DIDs storage ==

[[File:didstorage.png|thumb|400px]]

In the Admin/Settings page is possible to choose to use the DIDs storage. 
In this way the DIDs needs to be entered by the Admin/DIDs List menu and can be later chosen using a drop down box from the Define DID page. 
DIDs cannot be deleted, but they are just not assigned to any tenant.

== Outbound CallerID Regex ==

[[File:calleridregex.png|thumb|400px]]

DIDs are used by extensions as CallerID, but it is possible to select a Regex to pickup the most useful CallerID for that call, so you may use your italian office CallerID when dialing to Italy or your american one when dialing to US.

If your profile allows you to use customized CallerID, you can enter the CallerID number of your choice.

DIDs

2016-06-18T18:44:41Z

Manager: /* Bulk DIDs creation */

DIDs

2016-06-18T18:44:12Z

Manager: /* SMS */

[[File:Did.png|thumb|400px]]
To every tenant can be assigned a group of DIDs, numbers to be dialed from the outside. 
Each DID can be configured to reach a certain extension or Queue or Hunt Group or IVR or any combination of the elements defined for the tenant. 
The same DID cannot be assigned to more than one tenant and there is no correlation between the dialing profile assigned to a tenant, the trunk provider used and the incoming DID. 
For this reason is highly important that the incoming INVITE from the trunk provider contains all the info to identify correctly the DID it is addressing to. 

Due to the fact there is a lack of strong standardization among trunk providers on how to transmit the DID information, a “best guess” is used to identify the correct DID, using any of the format currently in use around the world.

A DID can be configured to accept voice, fax or try to guess the calling party (if voice of fax). 
Autodetection relies on signalling at the start of the call and cannot be always reliable. 
The time to detect the fax tone can be set in Admin/Settings page.

== Editing New/Define DIDs ==

[[File:newDID.png|thumb|400px]]

The number must be entered in three parts, starting with international prefix, area code and number. Even if your telephone standard doesn't allow the usage of any “short” form for dialing the number, requesting for example, to dial always the area code, the number is requested to be entered this way. Using the Admin/Settings menu is possible to enable the use of a “single box DID” to just enter the number in a single box, but this is highly discouraged. 
It is possible to use regular expression inside the number. In case multiple regular expressions match the number dialed, the ordering is based alphabetically on the Comment. 
'''Comment''' is just a comment and it is not used in any way. 
'''Max channels''' allows you to set the maximal number of channels available on the DID. Any additional call received will be refused with a busy signal. 
'''Use CNAM service''' allows you to assign the Caller ID Name based on this popular service offered in most countries. You can configure the service in Admin/Settings page. 
'''Use as Emergency CallerID''' permits you to mark this DID as one of the available to be used when an emergency call is dialed. Due to the fact the emergency numbers are not standard among all countries, it is your duty to assign the “emergency flag” to the dialing rules for emergency numbers. When a call is dialed and it is using a dialing rule marked as “emergency”, the Caller ID number used for the extension is the one defined as “Emergency” among the ones with this flag set. 
'''Inbound Call Rate''' allows you to define a call rate for inbound calls.

== Voice ==

This section allows to define how to route the call when a voice call is received. 
'''Unconditional Forward''' is a destination that can be set on the DID to send the call to a particular destination. It is enable/disabled by the checkbox and that checkbox can be easily controlled by a feature code. This is not the destination to be set for common usage, you need to use the one in the Voice section. 
'''Always record''' permits to force the recording of the call, whichever is the future of the call. Recordings will be available through the Call History menu. 
'''Email recordings''' to defines if and to who the recordings needs to be sent once the call is over. Multiple destinations email can be entered with any delimiter, like space, comma and point and comma. 
'''Minimum Size (Bytes)''' allows to receive those recordings bigger than the size set, in Bytes. Recording takes place only on bridged channels, so IVR prompts or Music On Hold will be not recorded. 
Using '''the Prefix CallerID Num''' is possible to define a string to be added to any CallerID number received to identify for example the DID the call is coming from. The same result can be achieved by defining a Custom Destination. 
'''Prefix CallerID Name''' is the same as above, but for the Caller ID Name. 
'''Destination multiple drop down''' allow to define the list and the order of the objects receiving the call.

== Fax ==

This section allows to define what to do with the call if a fax is received on the number defined. Receiving Fax over the Internet has often a low success rate, even if T.38 protocol is used.

[[File:didfax.png|thumb|400px]]

'''Receive fax''' can be used to select if autodetect, force or disallow the reception of a Fax over the current DID.

'''Fax Station ID''' and '''Fax Header''' allows you to customize your virtual fax.

'''Fax Protocol''' can be selected among T.38 (the preferred way to receive faxes), T.38 with fallback to G.711 or only G.711.

'''Email destinations box''' permits to enter one or multiple comma delimited email address to forward the received fax. Received fax are sent in PDF format. Partially received faxes are sent in the same way.

It is possible to store the fax received for later reviewing using the '''Store Fax Received control'''.

== SMS ==

[[File:sms.png|thumb|400px]]

You can specify how to pull or receive SMS on this DID.

'''Protocol''' lets you choose the protocol to be used. More info can be found in embedded help.

The SMS can be then delivered to a SIP phone as SIP message or to any other destination.

== Bulk DIDs creation ==

[[File:Bulkdid.png|thumb|400px]]

It is possible to create multiple DIDs at once with Bulk DIDs creation button. 
In this way a range of number is requested. 
All the generated numbers are configured in the same way.

== Use DIDs storage ==

[[File:didstorage.png|thumb|400px]]

In the Admin/Settings page is possible to choose to use the DIDs storage. 
In this way the DIDs needs to be entered by the Admin/DIDs List menu and can be later chosen using a drop down box from the Define DID page. 
DIDs cannot be deleted, but they are just not assigned to any tenant.

== Outbound CallerID Regex ==

[[File:calleridregex.png|thumb|400px]]

DIDs are used by extensions as CallerID, but it is possible to select a Regex to pickup the most useful CallerID for that call, so you may use your italian office CallerID when dialing to Italy or your american one when dialing to US.

DIDs

2016-06-18T18:43:51Z

Manager: /* SMS */

[[File:Did.png|thumb|400px]]
To every tenant can be assigned a group of DIDs, numbers to be dialed from the outside. 
Each DID can be configured to reach a certain extension or Queue or Hunt Group or IVR or any combination of the elements defined for the tenant. 
The same DID cannot be assigned to more than one tenant and there is no correlation between the dialing profile assigned to a tenant, the trunk provider used and the incoming DID. 
For this reason is highly important that the incoming INVITE from the trunk provider contains all the info to identify correctly the DID it is addressing to. 

Due to the fact there is a lack of strong standardization among trunk providers on how to transmit the DID information, a “best guess” is used to identify the correct DID, using any of the format currently in use around the world.

A DID can be configured to accept voice, fax or try to guess the calling party (if voice of fax). 
Autodetection relies on signalling at the start of the call and cannot be always reliable. 
The time to detect the fax tone can be set in Admin/Settings page.

== Editing New/Define DIDs ==

[[File:newDID.png|thumb|400px]]

The number must be entered in three parts, starting with international prefix, area code and number. Even if your telephone standard doesn't allow the usage of any “short” form for dialing the number, requesting for example, to dial always the area code, the number is requested to be entered this way. Using the Admin/Settings menu is possible to enable the use of a “single box DID” to just enter the number in a single box, but this is highly discouraged. 
It is possible to use regular expression inside the number. In case multiple regular expressions match the number dialed, the ordering is based alphabetically on the Comment. 
'''Comment''' is just a comment and it is not used in any way. 
'''Max channels''' allows you to set the maximal number of channels available on the DID. Any additional call received will be refused with a busy signal. 
'''Use CNAM service''' allows you to assign the Caller ID Name based on this popular service offered in most countries. You can configure the service in Admin/Settings page. 
'''Use as Emergency CallerID''' permits you to mark this DID as one of the available to be used when an emergency call is dialed. Due to the fact the emergency numbers are not standard among all countries, it is your duty to assign the “emergency flag” to the dialing rules for emergency numbers. When a call is dialed and it is using a dialing rule marked as “emergency”, the Caller ID number used for the extension is the one defined as “Emergency” among the ones with this flag set. 
'''Inbound Call Rate''' allows you to define a call rate for inbound calls.

== Voice ==

This section allows to define how to route the call when a voice call is received. 
'''Unconditional Forward''' is a destination that can be set on the DID to send the call to a particular destination. It is enable/disabled by the checkbox and that checkbox can be easily controlled by a feature code. This is not the destination to be set for common usage, you need to use the one in the Voice section. 
'''Always record''' permits to force the recording of the call, whichever is the future of the call. Recordings will be available through the Call History menu. 
'''Email recordings''' to defines if and to who the recordings needs to be sent once the call is over. Multiple destinations email can be entered with any delimiter, like space, comma and point and comma. 
'''Minimum Size (Bytes)''' allows to receive those recordings bigger than the size set, in Bytes. Recording takes place only on bridged channels, so IVR prompts or Music On Hold will be not recorded. 
Using '''the Prefix CallerID Num''' is possible to define a string to be added to any CallerID number received to identify for example the DID the call is coming from. The same result can be achieved by defining a Custom Destination. 
'''Prefix CallerID Name''' is the same as above, but for the Caller ID Name. 
'''Destination multiple drop down''' allow to define the list and the order of the objects receiving the call.

== Fax ==

This section allows to define what to do with the call if a fax is received on the number defined. Receiving Fax over the Internet has often a low success rate, even if T.38 protocol is used.

[[File:didfax.png|thumb|400px]]

'''Receive fax''' can be used to select if autodetect, force or disallow the reception of a Fax over the current DID.

'''Fax Station ID''' and '''Fax Header''' allows you to customize your virtual fax.

'''Fax Protocol''' can be selected among T.38 (the preferred way to receive faxes), T.38 with fallback to G.711 or only G.711.

'''Email destinations box''' permits to enter one or multiple comma delimited email address to forward the received fax. Received fax are sent in PDF format. Partially received faxes are sent in the same way.

It is possible to store the fax received for later reviewing using the '''Store Fax Received control'''.

== SMS ==

[[File:sms.png|thumb|400px]]

You can specify how to pull or receive SMS on this DID.
'''Protocol''' lets you choose the protocol to be used. More info can be found in embedded help.
The SMS can be then delivered to a SIP phone as SIP message or to any other destination.

== Bulk DIDs creation ==

[[File:Bulkdid.png|thumb|400px]]

It is possible to create multiple DIDs at once with Bulk DIDs creation button. 
In this way a range of number is requested. 
All the generated numbers are configured in the same way.

== Use DIDs storage ==

[[File:didstorage.png|thumb|400px]]

In the Admin/Settings page is possible to choose to use the DIDs storage. 
In this way the DIDs needs to be entered by the Admin/DIDs List menu and can be later chosen using a drop down box from the Define DID page. 
DIDs cannot be deleted, but they are just not assigned to any tenant.

== Outbound CallerID Regex ==

[[File:calleridregex.png|thumb|400px]]

DIDs are used by extensions as CallerID, but it is possible to select a Regex to pickup the most useful CallerID for that call, so you may use your italian office CallerID when dialing to Italy or your american one when dialing to US.

DIDs

2016-06-18T18:43:26Z

Manager: /* SMS */

[[File:Did.png|thumb|400px]]
To every tenant can be assigned a group of DIDs, numbers to be dialed from the outside. 
Each DID can be configured to reach a certain extension or Queue or Hunt Group or IVR or any combination of the elements defined for the tenant. 
The same DID cannot be assigned to more than one tenant and there is no correlation between the dialing profile assigned to a tenant, the trunk provider used and the incoming DID. 
For this reason is highly important that the incoming INVITE from the trunk provider contains all the info to identify correctly the DID it is addressing to. 

Due to the fact there is a lack of strong standardization among trunk providers on how to transmit the DID information, a “best guess” is used to identify the correct DID, using any of the format currently in use around the world.

A DID can be configured to accept voice, fax or try to guess the calling party (if voice of fax). 
Autodetection relies on signalling at the start of the call and cannot be always reliable. 
The time to detect the fax tone can be set in Admin/Settings page.

== Editing New/Define DIDs ==

[[File:newDID.png|thumb|400px]]

The number must be entered in three parts, starting with international prefix, area code and number. Even if your telephone standard doesn't allow the usage of any “short” form for dialing the number, requesting for example, to dial always the area code, the number is requested to be entered this way. Using the Admin/Settings menu is possible to enable the use of a “single box DID” to just enter the number in a single box, but this is highly discouraged. 
It is possible to use regular expression inside the number. In case multiple regular expressions match the number dialed, the ordering is based alphabetically on the Comment. 
'''Comment''' is just a comment and it is not used in any way. 
'''Max channels''' allows you to set the maximal number of channels available on the DID. Any additional call received will be refused with a busy signal. 
'''Use CNAM service''' allows you to assign the Caller ID Name based on this popular service offered in most countries. You can configure the service in Admin/Settings page. 
'''Use as Emergency CallerID''' permits you to mark this DID as one of the available to be used when an emergency call is dialed. Due to the fact the emergency numbers are not standard among all countries, it is your duty to assign the “emergency flag” to the dialing rules for emergency numbers. When a call is dialed and it is using a dialing rule marked as “emergency”, the Caller ID number used for the extension is the one defined as “Emergency” among the ones with this flag set. 
'''Inbound Call Rate''' allows you to define a call rate for inbound calls.

== Voice ==

This section allows to define how to route the call when a voice call is received. 
'''Unconditional Forward''' is a destination that can be set on the DID to send the call to a particular destination. It is enable/disabled by the checkbox and that checkbox can be easily controlled by a feature code. This is not the destination to be set for common usage, you need to use the one in the Voice section. 
'''Always record''' permits to force the recording of the call, whichever is the future of the call. Recordings will be available through the Call History menu. 
'''Email recordings''' to defines if and to who the recordings needs to be sent once the call is over. Multiple destinations email can be entered with any delimiter, like space, comma and point and comma. 
'''Minimum Size (Bytes)''' allows to receive those recordings bigger than the size set, in Bytes. Recording takes place only on bridged channels, so IVR prompts or Music On Hold will be not recorded. 
Using '''the Prefix CallerID Num''' is possible to define a string to be added to any CallerID number received to identify for example the DID the call is coming from. The same result can be achieved by defining a Custom Destination. 
'''Prefix CallerID Name''' is the same as above, but for the Caller ID Name. 
'''Destination multiple drop down''' allow to define the list and the order of the objects receiving the call.

== Fax ==

This section allows to define what to do with the call if a fax is received on the number defined. Receiving Fax over the Internet has often a low success rate, even if T.38 protocol is used.

[[File:didfax.png|thumb|400px]]

'''Receive fax''' can be used to select if autodetect, force or disallow the reception of a Fax over the current DID.

'''Fax Station ID''' and '''Fax Header''' allows you to customize your virtual fax.

'''Fax Protocol''' can be selected among T.38 (the preferred way to receive faxes), T.38 with fallback to G.711 or only G.711.

'''Email destinations box''' permits to enter one or multiple comma delimited email address to forward the received fax. Received fax are sent in PDF format. Partially received faxes are sent in the same way.

It is possible to store the fax received for later reviewing using the '''Store Fax Received control'''.

== SMS ==

You can specify how to pull or receive SMS on this DID.
'''Protocol''' lets you choose the protocol to be used. More info can be found in embedded help.
The SMS can be then delivered to a SIP phone as SIP message or to any other destination.

[[File:sms.png|thumb|400px]]

== Bulk DIDs creation ==

[[File:Bulkdid.png|thumb|400px]]

It is possible to create multiple DIDs at once with Bulk DIDs creation button. 
In this way a range of number is requested. 
All the generated numbers are configured in the same way.

== Use DIDs storage ==

[[File:didstorage.png|thumb|400px]]

In the Admin/Settings page is possible to choose to use the DIDs storage. 
In this way the DIDs needs to be entered by the Admin/DIDs List menu and can be later chosen using a drop down box from the Define DID page. 
DIDs cannot be deleted, but they are just not assigned to any tenant.

== Outbound CallerID Regex ==

[[File:calleridregex.png|thumb|400px]]

DIDs are used by extensions as CallerID, but it is possible to select a Regex to pickup the most useful CallerID for that call, so you may use your italian office CallerID when dialing to Italy or your american one when dialing to US.

DIDs

2016-06-18T18:43:02Z

Manager: /* SMS */

[[File:Did.png|thumb|400px]]
To every tenant can be assigned a group of DIDs, numbers to be dialed from the outside. 
Each DID can be configured to reach a certain extension or Queue or Hunt Group or IVR or any combination of the elements defined for the tenant. 
The same DID cannot be assigned to more than one tenant and there is no correlation between the dialing profile assigned to a tenant, the trunk provider used and the incoming DID. 
For this reason is highly important that the incoming INVITE from the trunk provider contains all the info to identify correctly the DID it is addressing to. 

Due to the fact there is a lack of strong standardization among trunk providers on how to transmit the DID information, a “best guess” is used to identify the correct DID, using any of the format currently in use around the world.

A DID can be configured to accept voice, fax or try to guess the calling party (if voice of fax). 
Autodetection relies on signalling at the start of the call and cannot be always reliable. 
The time to detect the fax tone can be set in Admin/Settings page.

== Editing New/Define DIDs ==

[[File:newDID.png|thumb|400px]]

The number must be entered in three parts, starting with international prefix, area code and number. Even if your telephone standard doesn't allow the usage of any “short” form for dialing the number, requesting for example, to dial always the area code, the number is requested to be entered this way. Using the Admin/Settings menu is possible to enable the use of a “single box DID” to just enter the number in a single box, but this is highly discouraged. 
It is possible to use regular expression inside the number. In case multiple regular expressions match the number dialed, the ordering is based alphabetically on the Comment. 
'''Comment''' is just a comment and it is not used in any way. 
'''Max channels''' allows you to set the maximal number of channels available on the DID. Any additional call received will be refused with a busy signal. 
'''Use CNAM service''' allows you to assign the Caller ID Name based on this popular service offered in most countries. You can configure the service in Admin/Settings page. 
'''Use as Emergency CallerID''' permits you to mark this DID as one of the available to be used when an emergency call is dialed. Due to the fact the emergency numbers are not standard among all countries, it is your duty to assign the “emergency flag” to the dialing rules for emergency numbers. When a call is dialed and it is using a dialing rule marked as “emergency”, the Caller ID number used for the extension is the one defined as “Emergency” among the ones with this flag set. 
'''Inbound Call Rate''' allows you to define a call rate for inbound calls.

== Voice ==

This section allows to define how to route the call when a voice call is received. 
'''Unconditional Forward''' is a destination that can be set on the DID to send the call to a particular destination. It is enable/disabled by the checkbox and that checkbox can be easily controlled by a feature code. This is not the destination to be set for common usage, you need to use the one in the Voice section. 
'''Always record''' permits to force the recording of the call, whichever is the future of the call. Recordings will be available through the Call History menu. 
'''Email recordings''' to defines if and to who the recordings needs to be sent once the call is over. Multiple destinations email can be entered with any delimiter, like space, comma and point and comma. 
'''Minimum Size (Bytes)''' allows to receive those recordings bigger than the size set, in Bytes. Recording takes place only on bridged channels, so IVR prompts or Music On Hold will be not recorded. 
Using '''the Prefix CallerID Num''' is possible to define a string to be added to any CallerID number received to identify for example the DID the call is coming from. The same result can be achieved by defining a Custom Destination. 
'''Prefix CallerID Name''' is the same as above, but for the Caller ID Name. 
'''Destination multiple drop down''' allow to define the list and the order of the objects receiving the call.

== Fax ==

This section allows to define what to do with the call if a fax is received on the number defined. Receiving Fax over the Internet has often a low success rate, even if T.38 protocol is used.

[[File:didfax.png|thumb|400px]]

'''Receive fax''' can be used to select if autodetect, force or disallow the reception of a Fax over the current DID.

'''Fax Station ID''' and '''Fax Header''' allows you to customize your virtual fax.

'''Fax Protocol''' can be selected among T.38 (the preferred way to receive faxes), T.38 with fallback to G.711 or only G.711.

'''Email destinations box''' permits to enter one or multiple comma delimited email address to forward the received fax. Received fax are sent in PDF format. Partially received faxes are sent in the same way.

It is possible to store the fax received for later reviewing using the '''Store Fax Received control'''.

== SMS ==

You can specify how to pull or receive SMS on this DID. 
'''Protocol''' lets you choose the protocol to be used. More info can be found in embedded help.
The SMS can be then delivered to a SIP phone as SIP message or to any other destination.

[[File:sms.png|thumb|400px]]

== Bulk DIDs creation ==

[[File:Bulkdid.png|thumb|400px]]

It is possible to create multiple DIDs at once with Bulk DIDs creation button. 
In this way a range of number is requested. 
All the generated numbers are configured in the same way.

== Use DIDs storage ==

[[File:didstorage.png|thumb|400px]]

In the Admin/Settings page is possible to choose to use the DIDs storage. 
In this way the DIDs needs to be entered by the Admin/DIDs List menu and can be later chosen using a drop down box from the Define DID page. 
DIDs cannot be deleted, but they are just not assigned to any tenant.

== Outbound CallerID Regex ==

[[File:calleridregex.png|thumb|400px]]

DIDs are used by extensions as CallerID, but it is possible to select a Regex to pickup the most useful CallerID for that call, so you may use your italian office CallerID when dialing to Italy or your american one when dialing to US.

DIDs

2016-06-18T18:41:58Z

Manager:

[[File:Did.png|thumb|400px]]
To every tenant can be assigned a group of DIDs, numbers to be dialed from the outside. 
Each DID can be configured to reach a certain extension or Queue or Hunt Group or IVR or any combination of the elements defined for the tenant. 
The same DID cannot be assigned to more than one tenant and there is no correlation between the dialing profile assigned to a tenant, the trunk provider used and the incoming DID. 
For this reason is highly important that the incoming INVITE from the trunk provider contains all the info to identify correctly the DID it is addressing to. 

Due to the fact there is a lack of strong standardization among trunk providers on how to transmit the DID information, a “best guess” is used to identify the correct DID, using any of the format currently in use around the world.

A DID can be configured to accept voice, fax or try to guess the calling party (if voice of fax). 
Autodetection relies on signalling at the start of the call and cannot be always reliable. 
The time to detect the fax tone can be set in Admin/Settings page.

== Editing New/Define DIDs ==

[[File:newDID.png|thumb|400px]]

The number must be entered in three parts, starting with international prefix, area code and number. Even if your telephone standard doesn't allow the usage of any “short” form for dialing the number, requesting for example, to dial always the area code, the number is requested to be entered this way. Using the Admin/Settings menu is possible to enable the use of a “single box DID” to just enter the number in a single box, but this is highly discouraged. 
It is possible to use regular expression inside the number. In case multiple regular expressions match the number dialed, the ordering is based alphabetically on the Comment. 
'''Comment''' is just a comment and it is not used in any way. 
'''Max channels''' allows you to set the maximal number of channels available on the DID. Any additional call received will be refused with a busy signal. 
'''Use CNAM service''' allows you to assign the Caller ID Name based on this popular service offered in most countries. You can configure the service in Admin/Settings page. 
'''Use as Emergency CallerID''' permits you to mark this DID as one of the available to be used when an emergency call is dialed. Due to the fact the emergency numbers are not standard among all countries, it is your duty to assign the “emergency flag” to the dialing rules for emergency numbers. When a call is dialed and it is using a dialing rule marked as “emergency”, the Caller ID number used for the extension is the one defined as “Emergency” among the ones with this flag set. 
'''Inbound Call Rate''' allows you to define a call rate for inbound calls.

== Voice ==

This section allows to define how to route the call when a voice call is received. 
'''Unconditional Forward''' is a destination that can be set on the DID to send the call to a particular destination. It is enable/disabled by the checkbox and that checkbox can be easily controlled by a feature code. This is not the destination to be set for common usage, you need to use the one in the Voice section. 
'''Always record''' permits to force the recording of the call, whichever is the future of the call. Recordings will be available through the Call History menu. 
'''Email recordings''' to defines if and to who the recordings needs to be sent once the call is over. Multiple destinations email can be entered with any delimiter, like space, comma and point and comma. 
'''Minimum Size (Bytes)''' allows to receive those recordings bigger than the size set, in Bytes. Recording takes place only on bridged channels, so IVR prompts or Music On Hold will be not recorded. 
Using '''the Prefix CallerID Num''' is possible to define a string to be added to any CallerID number received to identify for example the DID the call is coming from. The same result can be achieved by defining a Custom Destination. 
'''Prefix CallerID Name''' is the same as above, but for the Caller ID Name. 
'''Destination multiple drop down''' allow to define the list and the order of the objects receiving the call.

== Fax ==

This section allows to define what to do with the call if a fax is received on the number defined. Receiving Fax over the Internet has often a low success rate, even if T.38 protocol is used.

[[File:didfax.png|thumb|400px]]

'''Receive fax''' can be used to select if autodetect, force or disallow the reception of a Fax over the current DID.

'''Fax Station ID''' and '''Fax Header''' allows you to customize your virtual fax.

'''Fax Protocol''' can be selected among T.38 (the preferred way to receive faxes), T.38 with fallback to G.711 or only G.711.

'''Email destinations box''' permits to enter one or multiple comma delimited email address to forward the received fax. Received fax are sent in PDF format. Partially received faxes are sent in the same way.

It is possible to store the fax received for later reviewing using the '''Store Fax Received control'''.

== SMS ==

You can specify how to pull or receive SMS on this DID. 
'''Protocol''' lets you choose the protocol to be used. More info can be found in embedded help.
The SMS can be then delivered to a SIP phone as SIP message or to any other destination.

[[File:sms.png|thumb|400px]] 
 
 

[[File:sms2.png|thumb|400px]]

== Bulk DIDs creation ==

[[File:Bulkdid.png|thumb|400px]]

It is possible to create multiple DIDs at once with Bulk DIDs creation button. 
In this way a range of number is requested. 
All the generated numbers are configured in the same way.

== Use DIDs storage ==

[[File:didstorage.png|thumb|400px]]

In the Admin/Settings page is possible to choose to use the DIDs storage. 
In this way the DIDs needs to be entered by the Admin/DIDs List menu and can be later chosen using a drop down box from the Define DID page. 
DIDs cannot be deleted, but they are just not assigned to any tenant.

== Outbound CallerID Regex ==

[[File:calleridregex.png|thumb|400px]]

DIDs are used by extensions as CallerID, but it is possible to select a Regex to pickup the most useful CallerID for that call, so you may use your italian office CallerID when dialing to Italy or your american one when dialing to US.

File:Calleridregextop.png

2016-06-18T18:40:09Z

Manager:

Change the CallerID of the extension when dialing out

2016-06-18T18:39:56Z

Manager: Created page with "You may have the need to call out using different CallerID, some of them may be not even defined in the system. There are multiple ways to do it, let's review them: == Use Ca..."

Configuration Guides

2016-06-18T18:36:32Z

Manager:

FAQ

2016-06-18T17:39:19Z

Manager:

== Asterisk ==

; How can I setup Asterisk to use TLS : Asterisk can use TLS as transport for the signalling, increasing the authentication security and providing extra privacy about the number dialed and other info usually transmitted in clear over the SIP channel. TLS will encrypt only the signalling part, without offering any extra security to the RTP (voice) part. In the [general] section of sip.conf, add the following info, replacing the IP address with the IP address of the server. Provide the certificate and key in pem format.
<pre>tlsenable=yes
tlsbindaddr=213.133.102.85
tlscertfile=/etc/asterisk/certificates/demo.mirtapbx.com.pem
tlsdontverifyserver=no
tlscipher=DESCBC3SHA
tlsclientmethod=tlsv1
transport=udp,tls </pre>

== Asterisk Errors ==
; What is this error? <nowiki>WARNING[23261]: res_musiconhold.c:719 monmp3thread: poll() failed: Interrupted system call</nowiki> : Nothing to worry, when the Musing On Hold process terminates to play the media file, this error is thrown out, just ignore it

; What is this error? <nowiki>WARNING[3221]: func_cdr.c:352 cdr_write_callback: CDR requires a value (CDR(variable)=value)</nowiki> : Nothing to worry, it is a small glitch in the dialplan, but avoiding it will require an extra check, slowing down the call processing, so it is avoided, preferring the warning. Just ignore it.

; What is this error? <nowiki>ERROR[24984][C-00008457]: res_fax.c:4364 acf_faxopt_read: channel 'SIP/201-#######-0001547a' can't read FAXOPT(gateway) because it has never been written.</nowiki> : Nothing to worry, it is a small glitch in the dialplan, but avoiding it will require an extra check, slowing down the call processing, so it is avoided, preferring the error. Just ignore it.

; What is this error? <nowiki>WARNING[3221]: func_cdr.c:364 cdr_write_callback: Using the CDR function to set 'accountcode' is deprecated. Please use the CHANNEL function instead.</nowiki> : I am using an old syntax to preserve compatibility with older asterisk versions. Just ignore it.

; What is this error? <nowiki>WARNING[25144][C-0000845c]: dsp.c:1489 ast_dsp_process: Inband DTMF is not supported on codec #####. Use RFC2833</nowiki> : This is important, you are using an incompatible DTMF format for the codec you selected. Inband DTMF is supported in only a limited number of codec. Use RFC2833 as suggested.

; What is this error? <nowiki>NOTICE[9505] manager.c: 217.28.216.250 tried to authenticate with nonexistent user 'admin'</nowiki>
; What is this error? <nowiki>NOTICE[9505] manager.c: 217.28.216.250 failed to authenticate as 'admin'</nowiki> : Someone really stupid is trying to connect using the manager interface (port 5038). The manager interface is often locked down by IP address so it is really unlikely to be hacked this way, however this can be just annoying. To stop it after a few attempts, you can tweak the fail2ban to capture also these attempts by adding the following row in /etc/fail2ban/filters.d/asterisk.conf: '''<nowiki>^(%(__prefix_line)s|\[\]\s*)%(log_prefix)s <HOST> failed to authenticate as '[^']*'$</nowiki>''' and reload fail2ban

; What is this error? <nowiki>WARNING[26421][C-0000b22f]: chan_sip.c:7350 sip_write: Can't send 10 type frames with SIP write</nowiki> : Asterisk doesn't yet support comfort noise generation. Just ignore it.

; What is this error? <nowiki>NOTICE[6534] chan_sip.c: Received SIP subscribe for peer without mailbox: 226-SINI</nowiki> : The extension 226-SINI has sent a "SUBSCRIBE" message to asterisk, but the extension has no voicemail mailbox associated. This can be due to a configuration problem, you forget to associate the MWI mailbox in Configuration/Extension, or an asterisk problem... some time asterisk loads the extension without the mailbox associated. In this case you can restart the phone and deregister the extension, in this way when the phone registers again and subscribe to the voicemail mailbox, it should be loaded correctly. Otherwise you can move that extension to "No (Use externnotify)" in "Send MWI only if subscribed:" and set externnotify=/var/lib/asterisk/agi-bin/vmnotify.php in /etc/asterisk/voicemail.conf

; What is this error? <nowiki>WARNING[53344][C-000015c8] app_voicemail.c: SQL Get Data error! coltitle=category</nowiki> : This is a long standing asterisk bug fixed only in later releases... it is harmless and can be ignored

== Linux System ==

; When a conference is going to start, I get the message <nowiki>app_meetme.c:1296 build_conf: Unable to open DAHDI pseudo device</nowiki> : It seems the dahdi kernel module is not started or not compiled/available for your running kernel. It is possible you have upgraded your kernel and restarted your system. Try restarting the dahdi by using the command:

<pre>/etc/init.d/dahdi restart</pre>

If it doesn't fix the issue, try recompiling dahdi module, going in /usr/local/src/dahdilinux-complete-*
and running:

<pre>make
make install
/etc/init.d/dahdi restart
</pre>

; The web interface session is expiring too often, I need always to reauthenticate, how can I make it run longer? : You should change the session timeout value in php.ini and then restart the web server process, by default is 2880 seconds, set as long as you like

<pre>session.gc_maxlifetime = 2880</pre>

; How can I upgrade to PHP 5.5 to use AWS S3 Storage? : System is now shipped with PHP 5.5 already installed, but previous installations were using the standard PHP version which is not suitable for AWS S3 because it requires PHP 5.5. On CentOS 6 64bit You can upgrade with the following steps:

<pre>
rpm -Uvh https://mirror.webtatic.com/yum/el6/latest.rpm
yum -y install yum-plugin-replace
yum replace php-common --replace-with=php55w-common
yum install php55w-opcache
rpm --import https://mirror.webtatic.com/yum/RPM-GPG-KEY-webtatic-andy
service httpd restart
</pre>

; How can I change the server IP? : To change the server IP you need to refer to your OS documentation, but after you have changed your IP, there are a series of changes to be applied to MiRTA PBX:

* /etc/odbc.ini, /var/www/html/pbx/include/db.inc.php and /var/lib/asterisk/agi-bin/include/db.inc.php – contain the IP for the database server
* /etc/asterisk/sip.conf – contains the server itself definition used to allow the server to call itself. If you are running your server behind a NAT, it can be needed to change also the externip parameter. Once changed, you need to reload SIP (asterisk -rx 'sip reload'). Please note all extensions will be deregistered.
* /etc/asterisk/manager.conf – allows the web interface to access the manager interface. Once changed, you need to reload the manager interface (asterisk -rx 'manager reload')
* /etc/hosts – it is important to have the server host to resolv correctly to the new IP. Please don't use 127.0.0.1 as server IP
* /var/lib/asterisk/agi-bin/devstate.conf.php – lists the servers available in the pool to distribute the extension state. Status exchange is done over port 19771 using UDP protocol. Once the new IP has been inserted, you need to kill devstatesender.php and devstatereceiver.php processes, these will be automatically restarted.

Finally, you need to change the server IP defined in the web interface, using Admin/PBX Nodes. It can be a good idea to update the Always Allowed IP in the Admin/Security/GeoIP Fail2ban. It can be possible you need to request a new license for the server due to the change in IP. Please remember asterisk will still works and call will be processed normally even with an expired or not valid license.

== MySQL ==

; MySQL replication is broken with <nowiki>"Relay log read failure: Could not parse relay log event entry. The possible reasons are: the master's binary log is corrupted (you can check this by running 'mysqlbinlog' on the binary log), the slave's relay log is corrupted (you can check this by running 'mysqlbinlog' on the relay log), a network problem, or a bug in the master's or slave's MySQL code. If you want to check the master's binary log or slave's relay log, you will be able to know their names by issuing 'SHOW SLAVE STATUS' on this slave."</nowiki>, how can I fix it? : MySQL multimaster replication needs some advanced MySQL skills. If you do not have them, it is better if you refer to MiRTA PBX support and subscribe a "Nagios monitoring and OS management" service. If instead you have the needed knowledge, understanding that a wrong action can get things worst and you may lose data, to recover from this problem you need:

* Get the MySQL slave status with "show slave status"
* Stop the slave replication with "stop slave"
* Reset the relay log with "reset slave"
* Reinitialize the relay log with "change master to master_log_file='<Relay_Master_Log_File>', master_log_pos=<Exec_Master_Log_Pos>"
* Start the slave replication with "start slave"

== Extensions ==

; Can I use an extension to connect a remote PBX to the system? : Yes, but you need to enable the “trunk” feature for the extension or otherwise the Caller ID of the call coming from the remote PBX will be overwritten.

; When I use server side attended or unattended transfer (#* and ##), there is no enough time to dial the destination extension! : The default timeout is set to 3 seconds, but the “transfer” message is played inside this time, so it may seem shorter. You can increase the timeout of the transfer by editing the /etc/asterisk/features.conf and changing the value for transferdigittimeout to the amount of seconds you like. Once done, reload the module from within asterisk with “module reload features”

; When I try to recover the Voicemail, it says the PIN is invalid : Most of the time, there is a problem with DTMF, check the log for the call in /var/log/asterisk/full if you see this message: <code> dsp.c: Inband DTMF is not supported on codec g729. Use RFC2833</code> In this case, change the DTMF setting on the PBX from “auto” to RFC2833 and if possible, also on the phone

== Queues ==

; How can I get rid of the message "All of our representatives are currently..." : You need to set to 0 (zero) the Announce Frequency

== Voicemails ==

; A caller leaves a voicemail for an extension, but that voicemail is not appearing : The voicemail box can be locked, please check if in there a .lock file in the INBOX
folder, like in this example:
<pre>
#find /var/spool/asterisk/voicemail/pulmonarycriticl/100/
/var/spool/asterisk/voicemail/pulmonarycriticl/100/
/var/spool/asterisk/voicemail/pulmonarycriticl/100/Cust2
/var/spool/asterisk/voicemail/pulmonarycriticl/100/Urgent
/var/spool/asterisk/voicemail/pulmonarycriticl/100/INBOX
/var/spool/asterisk/voicemail/pulmonarycriticl/100/INBOX/.lock
/var/spool/asterisk/voicemail/pulmonarycriticl/100/Work
/var/spool/asterisk/voicemail/pulmonarycriticl/100/Cust3
/var/spool/asterisk/voicemail/pulmonarycriticl/100/Old
/var/spool/asterisk/voicemail/pulmonarycriticl/100/tmp
/var/spool/asterisk/voicemail/pulmonarycriticl/100/tmp/Av5Vqa.wav
/var/spool/asterisk/voicemail/pulmonarycriticl/100/tmp/Av5Vqa
</pre>
Just remove it. About the source of the lock… maybe your asterisk server has crashed in the middle of a voicemail message.

== Miscellaneous ==

; My clients are getting ghost calls from weird numbers not logged in MiRTA PBX : Those ghost calls are attempts made by “hackers” to place rogue calls, usually to premium rate numbers. They start by analyzing large part of Internet trying to connect to port 5060, the standard port used by PBX and phones. If they detect an answer, they try placing some calls using different formats. If one of these calls has success, then they start to send hundred of calls to premium rate numbers. They get some money rewards by phone companies. To avoid this issue, you can place the phone behind a firewall or NAT router, allowing only the PBX to connect or you can configure the phone to accept calls only from registered server. This option has several names depending by the phone brand.

; When a call has no callerid, is received as “asterisk”. How can I change it? : You can change setting the “callerid” parameter in the sip.conf and then reload sip from asterisk. Remember it will disconnect all clients connected.

; My Music on Hold is starting from the middle, but I want it to be started from the start each time : There are two ways to run Music on Hold... you can run a single process for all clients waiting in queue or you can start a new process for each of the clients. Obviously the first way is preferred if you have a big number of clients awaiting, but it has the drawback of having the MOH process to stream the music continuously, so a client joining the queue will start listening it not from the start. The second way can be activated by changing the musiconhold.conf and disabling cachertclasses

<pre>
[general]
cachertclasses=no ; use 1 instance of moh class for all users who are using it,
; decrease consumable cpu cycles and memory
; disabled by default
</pre>

This setting is PBX wide and cannot be turned on/off based on tenant

== Phones ==

; How can I avoid to receive a new call while I am already in a conversation? : You can receive another call while on line because the feature “Call Waiting” is active on your phone. Turn it off and the second call will get a BUSY signal. Disabling on the phone depends by the phone model, for example on Yealink it is here: [[File:yealink_callwaiting.png|thumb|400px]]

; How can I allow a SNOM phone to auto answer on paging? : SNOM has a security setting to prevent auto answer, so it has to be enabled to make it to work. On version V8 you need to go to Advanced / Behaviour (tab) / Phone Behaviour / Intercom policy [[File:snom_intercom.png|thumb|400px]]

== Provisioning ==

; My phone refuse to provision, but when I download the configuration, it seems perfect : If you are trying to provision over https, then check if the SSL certificate is valid. Phones require a valid certificate for provisioning. Verify the date and time on the phone because the certificate has a start and end date of validity. As last chance, try provisioning with http, but just to verify if the problem is in the certificate (some phones are really picky about certificates), then change the key and the extension password.

Provisioning

2016-06-17T20:51:46Z

Manager:

Provisioning of phones is phone independent, meaning MiRTA PBX has no knowledge of the various brands or models, but offers a generic framework to create a template and use some variables. Some variables are already stored in the system, but more can be added easily. Provisioning is made over HTTP and HTTPS. Obviously it is preferable to provision over HTTPS to avoid having SIP credentials to travel in clear over the Internet. Some brand of phones are quite picky about the certificate authority used for the SSL, so in case of problems, it can be good to make a test using HTTP or check the phone configuration about third party CA.

Some phone templates are provided as proof of concept, but they are not supposed to be neither complete neither accurate.

== Phone Models ==

Several phone models can be defined. For each phone model the number of lines needs to be defined, usually with the starting and ending line number. Please refer to the phone's provisioning guide to customize or create new templates.

[[file:phonemodes.png|thumb|400px]]

A phone template can be "locked" to prevent any accidental modification. You can easily duplicate a provisioning template using the action icon on the right

For each phone you can define a generic name, an option description, the first and last account number, the mac request template, the directory request template, the remote provisioning POST message along with the credentials to use for remote provisioning. Remote provisioning is the ability of some manufacturer to deliver a phone configuration just out of the box, like Yealink RPS and Polycom ZTP.

The template may contain variables in the general format ${name|default|type|options}.

Some variables are predefined and are read directly from the system configuration:

* line_active – Always 1
* displayname – Name of the Extension assigned
* virtualdisplayname – Name of the Virtual Extension assigned
* username – SIP username
* authname - SIP username
* number – Number of the Extension assigned
* virtualnumber – Number of the Virtual Extension assigned
* secret – Password of the Extension assigned
* index - The position of the account starting from the first line defined
* tenantcode - The account code for tenant
* subscribemwi - The option configured for "Send MWI only if subscribed"
* mailbox - The mailbox associated to the extension
* transport - The transport used for the extension (asterisk notation)
* codecs - The codecs configured for the extension (asterisk notation)
* line – Order position of the line (usually 1, 2, 3, etc)
* line_m1 - Order position of the line (usually 1, 2, 3, etc)
* key - Order position of the line (usually 1, 2, 3, etc)
* mac – Mac Address of the phone, without :
* provisionpassword – Password for provisioning URL

It is possible to define loops in the template using the {line_loop} and {/line_loop} keywords. Each variable used needs to be defined in the “Variables” menu. In case a variable is not
defined, an alert message will be shown. A default value can be defined for the variable, using the | character, like: {$server_host|demo.mirtapbx.com} To help users in configuring their system, you can specify the editing box type, so if by default is a text type as in:

Provisioning

2016-06-17T20:50:35Z

Manager:

Provisioning of phones is phone independent, meaning MiRTA PBX has no knowledge of the various brands or models, but offers a generic framework to create a template and use some variables. Some variables are already stored in the system, but more can be added easily. Provisioning is made over HTTP and HTTPS. Obviously it is preferable to provision over HTTPS to avoid having SIP credentials to travel in clear over the Internet. Some brand of phones are quite picky about the certificate authority used for the SSL, so in case of problems, it can be good to make a test using HTTP or check the phone configuration about third party CA.

Some phone templates are provided as proof of concept, but they are not supposed to be neither complete neither accurate.

== Phone Models ==

Several phone models can be defined. For each phone model the number of lines needs to be defined, usually with the starting and ending line number. Please refer to the phone's provisioning guide to customize or create new templates.

[[file:phonemodes.png|thumb|400px]]

A phone template can be "locked" to prevent any accidental modification. You can easily duplicate a provisioning template using the action icon on the right

For each phone you can define a generic name, an option description, the first and last account number, the mac request template, the directory request template, the remote provisioning POST message along with the credentials to use for remote provisioning. Remote provisioning is the ability of some manufacturer to deliver a phone configuration just out of the box, like Yealink RPS and Polycom ZTP.

The template may contain variables in the general format ${name|default|type|options}.

Some variables are predefined and are read directly from the system configuration:

* line_active – Always 1
* displayname – Name of the Extension assigned
* virtualdisplayname – Name of the Virtual Extension assigned
* username – SIP username
* authname - SIP username
* number – Number of the Extension assigned
* virtualnumber – Number of the Virtual Extension assigned
* secret – Password of the Extension assigned
* index - The position of the account starting from the first line defined
* tenantcode - The account code for tenant
* subscribemwi - The option configured for "Send MWI only if subscribed"
* mailbox - The mailbox associated to the extension
* transport - The transport used for the extension (asterisk notation)
* codecs - The codecs configured for the extension (asterisk notation)
* line – Order position of the line (usually 1, 2, 3, etc)
* line_m1 - Order position of the line (usually 1, 2, 3, etc)
* key - Order position of the line (usually 1, 2, 3, etc)
* mac – Mac Address of the phone, without :
* provisionpassword – Password for provisioning URL

Provisioning

2016-06-17T20:25:37Z

Manager:

File:Phonemodes.png

2016-06-17T20:22:38Z

Manager:

Provisioning

2016-06-17T20:22:23Z

Manager: Created page with "Provisioning of phones is phone independent, meaning MiRTA PBX has no knowledge of the various brands or models, but offers a generic framework to create a template and use so..."