Nove funkcije #14710
Zatvoren
gizmo/asterisk sigma-com officesa, podešenje asteriska, kvalitet call out-a
100%
Opis
podesiti gizmo account za asterisk
Povezani tiketi 2 (0 otvoreno — 2 zatvorenih)
Izmjenjeno od Ernad Husremović prije skoro 18 godina
How do I setup my Asterisk PBX to allow the Gizmo Project softphone to act as an extension number?¶
Solution This is an advanced topic. If you are new to Gizmo Project and VoIP, then you should probably just ignore this for now.
As you may be aware, one of the ways that Gizmo Project supports Asterisk, is by making it possible for the Gizmo Project softphone to act as an extension number on your Asterisk PBX, just like any other phone (or extension number) in the office.
Please note, that when properly setup, this will work, even when the user is not in your office. This means that even when someone is working from home (or on a business trip), they will still be able to send and receive calls on their office extension number, just like they would if they were sitting at their desk!
To do this, you will need:- An installed, and running Asterisk server. (Basic installation of Asterisk is beyond the scope of this article)
- A Fully Qualified Domain Name that can be used to access your Asterisk server over the Internet. (The actual connection will come from the Gizmo Project server, NOT from the Gizmo Project softphone)
You will need to create an extension number, and password for each user. Then create an entry for each user in the Asterisk "sip.conf" file, and modify the dialplan in the Asterisk "extensions.conf" file, to tell Asterisk to send calls to that extension to the associated SIP account.
New sip.conf settings:
[general] realm=YourDomain domain=YourDomain [UserExtension] type=friend ; allows incoming and outgoing calls username=UserExtension secret=UserPassword mailbox=UserExtension host=dynamic dtmfmode=rfc2833 canreinvite=yes allowguest=yes insecure=very promiscredir=yes
New extensions.conf settings:
[default] exten => UserExtension,1,Dial(SIP/UserExtension,130,t) ; tells Asterisk where to send the call when someone ; dials that extension number.
Izmjenjeno od Ernad Husremović prije skoro 18 godina
http://support.gizmoproject.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=402&nav=0
Connect Asterisk & Callweaver to Gizmo5 Sample Config:
Note: Callweaver is about the same except some additional built in STUN parameters.
File:sip.conf
[proxy01.sipphone.com] type=peer disallow=all allow=ulaw allow=ilbc dtmfmode=rfc2833 host=proxy01.sipphone.com fromdomain=proxy01.sipphone.com insecure=very qualify=yes fromuser=YOURSIP authuser=YOURSIP username=YOURSIP secret=YOURPASS canreinvite=no
Izmjenjeno od Ernad Husremović prije skoro 18 godina
evo kako stvar radi na našem asterisku:
postavke za call-out
[general] context=demo ;allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) canreinvite=no srvlookup=yes language=bs realm=asterisk.bring.out.ba domain=asterisk.bring.out.ba localnet=192.168.45.0/255.255.255.0 externhost=internet.sigma-com.net ;Specify how often (in seconds) a hostname DNS lookup should be performed for the value entered in 'externhost'. Default 10 seconds externrefresh=10 nat=yes register => 17473375695:ppppppwdddddddddddddd@proxy01.sipphone.com/17473375695 disallow=all allow=ulaw allow=alaw ;allow=ilbc allow=g729 qualify=yes [proxy01.sipphone.com] type=peer disallow=all allow=ulaw allow=alaw ;allow=ilbc dtfmode=rfc2833 host=proxy01.sipphone.com fromdomain=proxy01.sipphone.com insecure=very ; To allow registered hosts to call without re-authenticating qualify=yes fromuser=17473375695 authuser=17473375695 username=17473375695 secret=pppppppwddddddddd canreinvite=no nat=yes
postavka za podešenje gizmo klijenta u naš asterisk lan
[80] type=friend username=80 password=xxxxxxx host=dynamic callerid="gizmo_sigmacom 80" callgroup=3 pickupgroup=3 nat=yes qualify=yes disallow=all allow=ulaw allow=alaw ;allow=ilbc dtfmode=rfc2833 canreinvite=yes
gizmo se podešava tako da se logira na asterisk.bring.out.ba, user=80, password=pwd
Izmjenjeno od Ernad Husremović prije skoro 18 godina
kod prvog testa callout-a (zvao sašu na njegov 032 privatni broj) jasko je prijavio metalni zvuk
onda sam isključio ilbc codec (vidi se da je gore komentarisan) pa je taj metalni zvuk prestao
Izmjenjeno od Ernad Husremović prije skoro 18 godina
kod posljednjih testova imao sam prekide, a kada sam nazvao moju mamu, žalila se da me ne čuje - da joj prekidam
Izmjenjeno od Ernad Husremović prije skoro 18 godina
- Naslov promijenjeno iz gizmo/asterisk sigma-com officesa u gizmo/asterisk sigma-com officesa, podešenje asteriska, kvalitet call out-a
Izmjenjeno od Ernad Husremović prije više od 17 godina
- Status promijenjeno iz Dodijeljeno u Zatvoreno
- % završeno promijenjeno iz 0 u 100
kod svih ovih provajdera jedna stvar je bitna: minimalno tarifiranje je 1 minuta.
testirao sada poziv 70038732440170 - (officeze via gizmo, pozvao sa siemens gigaset-a) - sale kaže da je veza dobra. i kod mene je veza skroz čista, jedino što osjetim onaj silence supression kada ne razgovaramo
ja sam imao inače kredit nekih 7.9 $
evo gledam sada na mraka-2/vista ima instaliran gizmo klijent. cijena poziva je 0.171 $ a dužina poziva 0:47 - naplatio je 0.171$ gizmo
dobar je ovaj gizmo, kada sam ga tek postavio imao sam probleme sa registracijom na servere ali u zadnje vrijeme to provjerim sa onim echo call-om i to radi