The best document is the two page quick start guide that came in the box. You want 5.6, and 5.8 should be out soon if you are an early adopter.
-Jonathan Sent from a mobile device. On Dec 27, 2009, at 9:02 AM, Joseph <syscon...@gmail.com> wrote: > What what everybody says, it is a good hardware but configuration > samples are not easy to find and going through 500page manual is not > easy. > What they are missing is short configuration guide with samples for > specific software like asterisk. > My software version is 5.40A I see early next week what is the > latest available. > > On 12/27/09 07:56, Jonathan Thurman wrote: >> The web interface is a bit confusing at first. Here are some of the >> steps that I remember off hand. Change as little as possible, makes >> it easier to troubleshoot later. > > I did not change much and trying to register just one line first, > but is not easy all I'm getting is: > chan_sip.c:15593 handle_request_register: Registration from '<sip: > 3...@10.0.0.109>' failed for '10.0.0.157' - Wrong password > > 369 is my extension, 10.0.0.109 is my Asterisk server, 10.0.0.157 is > AudioCodes IP > >> >> Get the latest code from your vendor (5.6 is what I run) >> >> Configure the proxy to register with >> Configuration -> Protocol Config -> Protocol Def -> Proxy and >> Registration >> - Enable registration >> - Set the registration per endpoint > > So I have > Use Default Proxy: Yes > Proxy Set Table: ==> What did you enter here (I enter: 10.0.0.109 > UDP; do I need to set: Enable Proxy Keep Alive?) > > Proxy Name: 10.0.0.109 > > The below two settings (what to put in there, setting from sip.conf: > eg.: but which one? > Registrar Name > Registrar IP Address > > Under: > Gateway Name (I entered asterisk IP) 10.0.0.109 > > Again below is: > User Name > Password > Not sure what to put in above. > >> >> Configure your call routing >> Configuration -> Protocol Config -> Routing Tables -> IP to Trunk >> Group > > Is above sections for routing calls to asterisk? > >> >> If you send a prefix for outgoing calls, you will need to configure >> that in the manipulation table too >> Configuration -> Protocol Config -> Manipulation tables -> Dest >> number IP to Tel > > No, I don't use prefixes they are dropped by asterisk; so I > configured single stage dialing under: > Advanced Applications -> FXO Settings -> Dialing Mode > >> >> Configure authentication >> Configuration -> Protocol Config -> Endpoint settings -> >> Authentication > > Here I entered authentication from one of my sip.conf entry: [369] > [369] ; outgoing/incoming call on fxs port > type=friend > host=dynamic > context=internal > secret=523 > username=369 > mailbox=369 > ;dtmfmode=rfc2833 > ;dtmfmode=inband > disallow=all > allow=ulaw > allow=alaw > canreinvite=yes > nat=no > callgroup=1 > pickupgroup=1 > >> >> Now the part that took me a while to find... >> >> Configure the Channel to phone number mapping: >> Configuration -> Protocol Config -> Endpoint Number -> EndPoint >> Phone Number >> >> Configure the Hunt group settings >> Configuration -> Protocol Config -> Hunt/IP Group -> Hunt group >> settings >> >> >> Hope that helps. These are great devices, once you figure out how to >> get them configured... >> >> -Jonathan > > I need to find out from the manual what these setting do. > I was hoping to find some setting reference on Wiki but there are > none :-/ it seems to me the device is not very popular among > asterisk users, if it was > somebody would create detailed configuration for asterisk. > > -- > Joseph > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
