Asterisk settings Gizmo

Gizmo's Preferences

The Gizmo client chooses acceptable codecs based on the user's available bandwith. At a minimum, you will need to accept iLBC and GSM in order to talk to Gizmo users. On higher speed connections, it may be possible to use ulaw and alaw. Officially, Gizmo only supports iLBC and GSM (and strongly prefers GSM when possible).

Inbound SIP Calls

This used to be problematic due to Gizmo sending DTMF in such a way that it was incompatible with the codecs they were using — they're now using RFC2833. It should just be a matter of proper sip.conf entries. Disallow all codecs, allow GSM and iLBC.

Router notes:
Make sure you enable UPnP on your router. No outbound audio is a symptom of not doing so.

Firewall notes:

the standard SIP port (5060) does not suffice. we had to open the UDP port 5004 both destination an source i.e.


permit udp any eq 5004 host asterisk
permit udp any host asterisk eq 5004

sip.conf:

register => 1747xxxxxxx@proxy01.sipphone.com

[proxy01.sipphone.com]
type=friend
context=default
disallow=all
allow=ulaw
dtmfmode=rfc2833
host=proxy01.sipphone.com
insecure=very
secret=xxxx
username=<sipphone 10 digit phone #>
canreinvite=no


Really, gizmo (sipphone) support more codecs, like GSM and others, so it is not neccessary to enable only ulaw codec.
Usually, gizmo users known only their own username (alphanumerical), not the phone number associate to the username. To know your 10 digit phone number, just view your profile in the gizmo project program.

extensions.conf:

[default]
exten => s,1,dial(SIP/205)




Outbound SIP Calls

Outbound's much easier. You just need to dial the full 11-digit subscriber ID at proxy01.sipphone.com. We'll setup a context called to_gizmo below o facilitate this. All Gizmo and SIPPhone customers are assigned to the 747 or 222 area codes. When dialing, you must include the '1' (ie - 1747XXXXXXX) in order to reach the user.

Add to extensions.conf:

[to_gizmo]
exten => _91747.,1,SetCallerID("Your Name" <Your Num>)
exten => _91747.,2,Dial(SIP/${EXTEN:1}@proxy01.sipphone.com,20,r)
exten => _91747.,3,Congestion(5)
exten => _91222.,1,SetCallerID("Your Name" <Your Num>)
exten => _91222.,2,Dial(SIP/${EXTEN:1}@proxy01.sipphone.com,20,r)
exten => _91222.,3,Congestion(5)
exten => _49X.,1,Dial(SIP/${EXTEN:2}@proxy01.sipphone.com,20,r)


Drop the nines, of course, if you want to direct-dial rather than dialing for an outside line.
Note: Be sure to leave the < > in place in the above example.
The last line permit you to call every sipphone user and service, like "**" which permit you to know your sip number: just dial 49** to do that, or to call another number, dial 49NUMBER



Registering with Gizmo's Servers

If you want to receive calls to your Gizmo number using your Asterisk system, add an appropriate register line to your sip.conf. Do note that passwords are case-sensitive! Your password must be entered completely in lowercase, regardless of how you typed it when you signed up with Gizmo Project. (This is also true of the Gizmo Project web site, if you've had problems logging in there.)

Use your 10-digit Gizmo number and your Gizmo password, registering to proxy01.sipphone.com.

For example:
register => 17475551212:password@proxy01.sipphone.com

This is wrong:
register => 17475551212:PaSSworD@proxy01.sipphone.com


Gizmo is automatically configured with Intuitive Voice's Evolution PBX

Using Gizmo with Asterisk@Home 1.5

See also


Gizmo's Preferences

The Gizmo client chooses acceptable codecs based on the user's available bandwith. At a minimum, you will need to accept iLBC and GSM in order to talk to Gizmo users. On higher speed connections, it may be possible to use ulaw and alaw. Officially, Gizmo only supports iLBC and GSM (and strongly prefers GSM when possible).

Inbound SIP Calls

This used to be problematic due to Gizmo sending DTMF in such a way that it was incompatible with the codecs they were using — they're now using RFC2833. It should just be a matter of proper sip.conf entries. Disallow all codecs, allow GSM and iLBC.

Router notes:
Make sure you enable UPnP on your router. No outbound audio is a symptom of not doing so.

Firewall notes:

the standard SIP port (5060) does not suffice. we had to open the UDP port 5004 both destination an source i.e.


permit udp any eq 5004 host asterisk
permit udp any host asterisk eq 5004

sip.conf:

register => 1747xxxxxxx@proxy01.sipphone.com

[proxy01.sipphone.com]
type=friend
context=default
disallow=all
allow=ulaw
dtmfmode=rfc2833
host=proxy01.sipphone.com
insecure=very
secret=xxxx
username=<sipphone 10 digit phone #>
canreinvite=no


Really, gizmo (sipphone) support more codecs, like GSM and others, so it is not neccessary to enable only ulaw codec.
Usually, gizmo users known only their own username (alphanumerical), not the phone number associate to the username. To know your 10 digit phone number, just view your profile in the gizmo project program.

extensions.conf:

[default]
exten => s,1,dial(SIP/205)




Outbound SIP Calls

Outbound's much easier. You just need to dial the full 11-digit subscriber ID at proxy01.sipphone.com. We'll setup a context called to_gizmo below o facilitate this. All Gizmo and SIPPhone customers are assigned to the 747 or 222 area codes. When dialing, you must include the '1' (ie - 1747XXXXXXX) in order to reach the user.

Add to extensions.conf:

[to_gizmo]
exten => _91747.,1,SetCallerID("Your Name" <Your Num>)
exten => _91747.,2,Dial(SIP/${EXTEN:1}@proxy01.sipphone.com,20,r)
exten => _91747.,3,Congestion(5)
exten => _91222.,1,SetCallerID("Your Name" <Your Num>)
exten => _91222.,2,Dial(SIP/${EXTEN:1}@proxy01.sipphone.com,20,r)
exten => _91222.,3,Congestion(5)
exten => _49X.,1,Dial(SIP/${EXTEN:2}@proxy01.sipphone.com,20,r)


Drop the nines, of course, if you want to direct-dial rather than dialing for an outside line.
Note: Be sure to leave the < > in place in the above example.
The last line permit you to call every sipphone user and service, like "**" which permit you to know your sip number: just dial 49** to do that, or to call another number, dial 49NUMBER



Registering with Gizmo's Servers

If you want to receive calls to your Gizmo number using your Asterisk system, add an appropriate register line to your sip.conf. Do note that passwords are case-sensitive! Your password must be entered completely in lowercase, regardless of how you typed it when you signed up with Gizmo Project. (This is also true of the Gizmo Project web site, if you've had problems logging in there.)

Use your 10-digit Gizmo number and your Gizmo password, registering to proxy01.sipphone.com.

For example:
register => 17475551212:password@proxy01.sipphone.com

This is wrong:
register => 17475551212:PaSSworD@proxy01.sipphone.com


Gizmo is automatically configured with Intuitive Voice's Evolution PBX

Using Gizmo with Asterisk@Home 1.5

See also


Created by: colinm, Last modification: Fri 19 of Dec, 2008 (01:42 UTC) by vonreut
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