Asterisk Phone Bria

How to configure Asterisk and CounterPath Bria

Bria is the newest softphone application from CounterPath, allowing users to enjoy multimedia communications in a dynamic way.

Featuring an intuitive interface, Bria is expanding the softphone experience by making it even easier to make VoIP and Video over IP calls, see when your contacts are available and send Instant Messages.

While the popular eyeBeam 1.5 softphone uses a familiar-looking dialpad-centric user interface (which looks and feels like a regular phone), Bria features a contact-centric interface which focuses on your address book. This allows you to manage your communications more conveniently and according to your personal preferences.

Bria account configuration:

Menu | File | Account Settings

Create an account for your Asterisk connection with the following settings:

Account page

Account Name: <Full Name>
Protocol: SIP
User ID: <username>@x.x.x.x
Password: <secret>
Display Name: <Full Name>
Authorization name: <username>
Register with Domain and receive calls: CHECKED
Send outbound via:

Voicemail page (optional)

Check for Voicemail: CHECKED
Number to dial for checking voicemail: <internal number to dial for voicemail control mode>
Number for sending calls to voicemail: <internal number to dial for voicemail>

Bria configuration:

Make sure that Silence Supression (to avoid RFC3389 warnings on Asterisk console) is diabled:
Menu | File | Preferences | Network
Preserve bandwidth during silence periods: UNCHECKED

There is another serious issue using Bria with Asterisk 1.14: If you place a call on hold, the call will be terminated automatically after 30 seconds. The reason for this is explained in this posting in the Asterisk Mailing List:

Bria has an inactivity timer that is based on RTCP. Basically, if during the call there is RTCP, Bria uses it to make sure the call is still alive. Asterisk does send RTCP when call is active, but it stops when call is put on hold by Bria. The default timeout for Bria is 30 seconds, thus it disconnects the call because it has not received any RTP or RTCP during this time.

In order to solve this issue, a very internal setting in Bria must be modified. Unfortunately these settings are not accessible from the normal preferences dialog, so you have to use a little trick in order to access the hidden settings.

To access the expert settings mode, dial the following sequence: ***7469

Now a wonderful Expert-Mode-Settings-Dialog will appear, displaying a plethora of settings. Let's search the correct ones. Enter rtcp in the Find field. Now you will see the following settings:

Setting Value
rtp:inactivity:must_have_rtcp 1
rtp:inactivity:rtcp_timer_in_ms 30000
rtp:rtcp:frequency_to_send_in_ms 3000
rtp:send_rtcp_early_on_problem:enabled 1
rtp:send_rtcp_early_on_problem:min_packets_sent_by_remote_party 100
rtp:send_rtcp_early_on_problem:min_percentage_lost 5

Now you have two different possibilities to solve your problem:

  1. You modifiy the the value of the RTCP-Timer to something huge, so that the timeout will not occur any more after 30 seconds. To do this, change the value of rtp:inactivity:rtcp_timer_in_ms to let's say 300000. This means 300 seconds (5 minutes) instead on 30 seconds. This should be sufficient.
  2. You disable completely the RTCP-Timer feature. To do so, you set the value of rtp:inactivity:must_have_rtcp to 0

It may be also a good idea to disable completely the RTP inactivity timer. To do so, enter rtp:inactivity in the Find field. Now you will see the following settings:

Setting Value
rtp:inactivity:must_have_rtcp 1
rtp:inactivity:rtcp_timer_in_ms 30000
rtp:inactivity:rtp_timer_in_ms 0
rtp:inactivity:timer_enabled 1
rtp:inactivity:timer_in_ms 300000

To do so, you set the value of rtp:inactivity:timer_enabled to 0

In order to have your new settings active, you have to quit Bria (You have to select EXIT from the tray menu) and restart it. Now calls on hold are not terminated any more after 30 seconds.

Asterisk configuration:

Standard SIP user in sip.conf & extensions.conf

Example Configuration in sip.conf

; These templates are used to shorten up the phone entries and describe the
; model specific technical part
; This is the template for all eyebeam phones (also knows as XLite/XPro)

; These templates are used to shorten up the phone entries and describe the
; port type specific configuration
; This is the template for internal lines

callerid="Leopoldo's Computer" <10>

callerid="Leopoldo's Laptop" <12>

Asterisk | Asterisk Configuration | Channel Configuration | Configuration for Specific Phones
Created by: Tuxpoldo, Last modification: Wed 11 of Jun, 2008 (12:05 UTC)
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