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:[email protected]>' 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
