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Configuration Examples

  1. Rating: +2

    Positive Negative

    Sample Asterisk 1.6 Configuration

    The following configuration works for Asterisk/Trixbox 1.6:

    on sip.conf

    [GRNVoIP]
    type=peer
    dtmfmode=rfc2833
    promiscredir=yes
    canreinvite=no
    username=admin console username
    secret=admin console password
    fromdomain=brandid.grnvoip.com
    host=sbc.ezcallinc.com
    outboundproxy=sbc.ezcallinc.com
    insecure=port,invite
    disallow=all
    allow=g729
    allow=g723
    allow=ulaw
    allow=alaw

    *if md5secret is preferred, replace secret with the md5 hash value, eg: md5secret=bd405bf86c4cf427f7ca5466668961e7, where md5secret is md5 of username:realm:password

    on extensions.conf (to set a fix CLI)

    [grnvoip]
    exten => _Z.,1,NoOP
    exten => _Z.,2,Set(CALLERID(number)=1234567890)
    exten => _Z.,3,NoOP($(CALLERID(number))
    exten => _Z.,n,Dial(SIP/999999${EXTEN}@GRNVoIP,,R)
    exten => _Z.,n,Hangup

    on extensions.conf (for asterisk to pass through CLI)

    [grnvoip]
    exten => _Z.,1,NoOP
    exten => _Z.,n,Dial(SIP/999999${EXTEN}@GRNVoIP,,R)
    exten => _Z.,n,Hangup

    Note: replace 999999 with the prefix assigned to you for Standard, Premium or Special rates. If you are unsure of this, please contact Support for assistance.

  2. Rating: -2

    Positive Negative

    Do I need a SIP Username and Password?

    By default, grnVoIP uses IP address authentication only. However, If your equipment is situated behind NAT or if you are using a SIP domain name for authentication, you will be required to configure a SIP username and password in your equipment.

    (a) The SIP username and password is based on your login credentials to the web Management Console, but does not use the domain portion of your username. For example, if your web username is abacus123@acme.com, then your SIP username is "abacus123".

    (b) The SIP password will remain the same as your web login password. If you change your web login password, you must also change your SIP password.

  3. Rating: +3

    Positive Negative

    Basic Configuration Guidelines

    When sending traffic to grnVoIP.com, your gateways should be configured as follows:

    Your originating gateways should send traffic to our session border controller at sip:sbc.ezcallinc.com. If your device does not support a domain, please try the IP address 70.42.72.49.

    Our session border controller will accept both SIP and H.323 traffic and is a signaling proxy only. We use the standard ports - 5060 for SIP and 1720 for H.323.

    All interconnected gateways should be configured to use RFC2833 for DTMF relay.

    For H.323 Gateways, FASTSTART must be enabled.

    Our routing is based on e164 addresses with a unique prefix. Therefore, please send the destination number in the format of: PREFIX+COUNTRY CODE+CITY(or AREA) CODE+NUMBER

  4. Rating: +2

    Positive Negative

    Sample dialplan to insert prefix for Counterpath/X Lite/Bria softphone

    It is possible to add the dialing prefix to calls from Xlite. For example, if you would like to send the prefix 11780414 by default for any number input that starts with a "+", use this dial plan...

    +x.T;match=1;pre="11780414"

    Be sure to replace the sample prefix above with the actual prefix that was assigned to you. This was contaned in an email with the subject "Getting Started..."

    For more examples of dial plans, see the Xlite manual here:

    http://www.counterpath.com/assets/files/191/X-Lite3.0_UserGuide.pdf

  5. Rating: -2

    Positive Negative

    Sample dial peer for Cisco voice router.

    voice class codec 100
    codec preference 1 g729r8
    codec preference 2 g729br8
    codec preference 3 g723r63
    codec preference 4 g723ar63
    codec preference 5 g723r53
    codec preference 6 g723ar53

    dial-peer voice 10000 voip
    description EXAMPLE.DOMAIN.COM
    destination-pattern 1T
    tech-prefix 19514200
    voice-class codec 100
    voice-class sip outbound-proxy ipv4:sbc.domain.com
    session protocol sipv2
    session target dns:example.domain.com
    session transport udp
    dtmf-relay rtp-nte

    NOTES:

    1. Be sure to replace the domain example.domain.com with the actual domain that was assigned to you upon signup.

    2. Also replace the example tech-prefix of 19514200 with the actual tech-prefix that was assigned to you upon sign-up.

  6. Rating: +3

    Positive Negative

    Freeswitch configuration example

    The following steps will allow you to route US (1+) calls via grnVoIP:

    1. Add the following lines within the context from default.xml (where default is the profile used)

    $freeswitch/conf/dialplan/default.xml

    {extension name="grnvoip"}
    {condition field="destination_number" expression="^(1{0,1}d{10})$"}
    {action application="set" data="hangup_after_bridge=true"/}
    {action application="bridge" data="sofia/external/12345$1@70.42.72.49"/}
    {action application="hangup"/}
    {/condition}
    {/extension}

    where 12345 is the tech prefix provided to you upon sign up

    2. Add the following lines within the settings from acl.conf.xml

    $freeswitch/conf/autoload_configs/acl.conf.xml

    * from freeswitch cli execute 'reloadxml' for changes to take effect.

    NOTE:
    - The settings above will route calls with the sample prefix 12345. Remember to replace the sample with the actual prefix that was assigned to you when you signed-up.

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