login | register
Tue 09 of Feb, 2010 [21:09 UTC]

voip-info.org

History

Asterisk settings Gizmo

Created by: colinm,Last modification on Fri 19 of Dec, 2008 [01:42 UTC] by vonreut

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



Comments

Comments Filter
222

333No Audio from called party

by pabut, Wednesday 05 of August, 2009 [02:01:22 UTC]
I have purchased ... and I belive setup correctly Gizmo Call Out on my asterisk box ..... I can orginate an outbound call but I cannot hear the CALLED party .... the CALLED PARTY hears me fine. Any thoughts on where I should be looking????

Inbound (to the asterisk box via GIZMO call in ) there is no issue at all .... both parties here each other.
222

333multiple incoming numbers to differing places in dialplan

by inetws, Saturday 26 of May, 2007 [20:55:29 UTC]
I am running Asterisk 1.4 and able to successfully receive Gizmo calls. I have 2 Gizmo Call in Numbers. I want to send callers to different places in the dialplan dependent upon which number they call.
Placing the following in the extensions.conf in the context defined in sip.conf for sip-gizmo phone
exten => _17035551111,1,Goto(menu1111,s,1);
exten => _17035552222,1,Goto(menu2222,s,1);
doesn't catch the calls since they are being viewed as coming from SIP/MYGIZMOPHONESIPNUMBER regardless of which number is called.
222

333Gizmo incoming calls to Asterisk 1.4.1

by collectiveb, Friday 16 of March, 2007 [13:29:23 UTC]
I have just upgraded my Asterisk server to version 1.4.1 and it no longer accepts incoming calls from my Gizmo account.<br>
My other provider SIP accounts had the same problem until I changed :-
insecure=very

to
insecure=port,invite

in each providers settings in sip.conf<br>
I have tried all the combinations of port and invite without success.<br>
(insecure=very is no longer supported in Asterisk 1.4.1)
Anyone else having incoming gizmo call problems ?<br>
Outgoing works perfectly btw.<br>
<br>
Posted on the Gizmo Forums here :-<br>
http://forum.gizmoproject.com/viewtopic.php?t=6180

222

333Gizmo incoming calls to Asterisk 1.4.1

by collectiveb, Friday 16 of March, 2007 [13:28:52 UTC]
I have just upgraded my Asterisk server to version 1.4.1 and it no longer accepts incoming calls from my Gizmo account.<br>
My other provider SIP accounts had the same problem until I changed :-
insecure=very

to
insecure=port,invite

in each providers settings in sip.conf<br>
I have tried all the combinations of port and invite without success.<br>
(insecure=very is no longer supported in Asterisk 1.4.1)
Anyone else having incoming gizmo call problems ?<br>
Outgoing works perfectly btw.<br>
<br>
Posted on the Gizmo Forums here :-<br>
http://forum.gizmoproject.com/viewtopic.php?t=6180

222

333using secondary account for Asterisk

by sshadow, Sunday 25 of June, 2006 [07:45:17 UTC]
Just to add a note: When trying to configure your secondary account in the gizmo softphone, be warned that your Asterisk server has to be available through the internet.
This means that the Asterisk server must have a valid public IP. See this <a href = "http://forum.gizmoproject.com/viewtopic.php?t=2438" >link </a>

<br>The Gizmo server will register your account on the Asterisk server.




222

333For SVN-trunk, username must match

by capouch, Wednesday 10 of May, 2006 [05:53:13 UTC]
I found that I had to make the "context" name (i.e. the part in the square brackets) match the username in order to get the registration to work--in other words, the context had to be called "proxy01.sipphone.com"

This is for the SVN-trunk versions of Asterisk. I don't run any of the stable ones to know if it would be a problem there, too.

Once I did that everything worked fine. When I had it like the example at the top of this thread (using the name gizmo-trunk) it did many massive weird things to the server.

222

333Not sure if this warrants a change to the page, but here is what worked for me (the above doesn't):

by snfettig, Thursday 16 of March, 2006 [18:45:27 UTC]
I found that the above did NOT work and had to take information from three different sources. sip.conf file should have the following:
<br>
gizmo_trunk<br>
allow=ulaw&alaw&ilbc<br>
context=from-pstn<br>
disallow=all<br>
dtmfmode=rfc2833<br>
fromdomain=proxy01.sipphone.com<br>
fromuser=1747XXXXXXX<br>
host=proxy01.sipphone.com<br>
insecure=very<br>
secret=your_password<br>
type=friend<br>
username=1747XXXXXXX<br>
<br>
register=1747XXXXXXX:your_password@proxy01.sipphone.com<br>
<br>
This allowed me to make outbound calls, as my account is set up for paid outbound dialing. I posted my settings for Asterisk@Home (v. 2.7) to <a href="http://stevenfettig.com/mythoughts/archives/000230.php">my weblog</a> which made inbound calls come through, too.
222

333Almost perfect

by tedharris, Monday 28 of November, 2005 [20:15:24 UTC]
Dial in and out works perfectly, but I always appear 'offline' to Gizmo clients.

Does anyone know how to send presence info from Asterisk to Gizmo?
222

333works!

by bbrendon, Thursday 17 of November, 2005 [07:23:48 UTC]
Gijs, finally figured it all out. try the above edits and you should be up and running.
222

333Re: Gizmo ' not allowed' ?

by bbrendon, Tuesday 18 of October, 2005 [19:08:30 UTC]
I have the same problem.
Got SIP response 488 "Not Acceptable Here" back from 198.65.166.131