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Thread ID: 124395	2012-04-24 07:12:00	Linksys SPA3102 or Alternatives	Geek4414 (12000)	Press F1

Post ID	Timestamp	Content	User
1271649	2012-05-02 02:32:00	Cool! Hopefully this should be a bit more straight-forward... Unfortunately, No. I started to create a VM from the Elastix ISO and it appears to be suffering from the same problem as the AsteriskNow ISO. It's obviously trying to start X server and a GUI but broke and cannot complete the rest of the install ... INIT: Id "x" respawning too fast: disabled for 5 minutes And it keeps coming up if you leave it. I can just press Enter and login as root (as I did with AsteriskNow). Looks like I will have to go through downloading X-server and gdm and Asterisk manually from this point on. Argggh, why doesn't it just work (don't tell me to go and get a Fruity product please).	Geek4414 (12000)
1271650	2012-05-02 03:26:00	Saw this in the Elastix forum (www.elastix.org) "I don't think the Easy install wizard should be used. You just create a virtual machine, and connect a CD/DVD drive (or if I remember correctly the ISO image (without having to burn it)), and let it install by itself." So here we go again ... just created a VM and manually booting from the ISO and re-installing Elastix from scratch now... we shall see how that goes.	Geek4414 (12000)
1271651	2012-05-02 11:46:00	After following the advice in the Elastix forum shown in my previous post. I created each of the VM first manually without using the easy install wizard, then boot the VM from the respective ISO and their installations worked properly! I now have ALL of them installed ... 3 separate VMs with AsteriskNow, Asterisk FreePBX, Elastix and 3CX installed on a Windows XP box. So far, purely from the aesthetic & ease of use point of view, I found 3CX is the easiest to follow. Elastix is a more comprehensive system with a nice GUI, but it might be a bit confusing by having too much. AsteriskNow has a nice and improved GUI than what I have used in the past. Asterisk FreePBX is very sparse and I find it the least easy to follow and less appealing visually.	Geek4414 (12000)
1271652	2012-05-02 12:01:00	Hi Chilling_Silence I now have ALL of them (AsteriskNow/FreePBX/Elastix/3CX) installed! But still not sure how to get the SPA talking properly to any of the VoIP systems. It's piece of cake to create extensions and log in from the softphones. With 3CX, it works over tunnelling easily over the internet as remote extensions, without having to VPN in. Managed to get the softphone extensions to call the analog phone attached to the SPA and vice versa in some config but only one way in the other configs. All I want to achieve is to divert ALL incoming calls from the analog phone line to an extension where people can leave voice mail, and hopefully record the caller ID. I can then pick up these voice mail while overseas by calling the voice mail extension from my softphones. I don't need any fancy filtering, just divert ALL incoming calls. Also to dial out via the PSTN to make local calls from my softphones (Android and/or laptop), so I can call family and clients everyday. All these worked great a few years ago when I was in Europe for two months. Just need that going again before I go overseas for a month. I think what stumbles me is "Trunk", I haven't got the concept of that yet. According to your friend's set up, "Line 1" is set as User ID 399; which means "extension" 399, right? My interpretation of the dial plan (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) is as follow ... *xx * button followed by two digits only 0 Single digit 0 00 Double digit 00 [3469]11 means 311, 411, 611 and 911 only [2-9]xxxxxx means single digit between 2 to 9 then followed by 6 other digits 1xxx[2-9]xxxxxxS0 means start with "1" followed by 3 digits, then follow by a digit between 2 to 9 then follow by 6 digits and no pause? xxxxxxxxxxxx. 12 digit numbers? the final "." is repetition? For the PSTN line, you have user ID set as PSTN and a password, is this then the "Trunk" and not an "extension"? So does Asterisk log into the SPA (instead of SPA registering extension on Asterisk)? Looks like all the dial plans on the PSTN tab are kept as default of (xx.) which is at least two digits and repetition (??)	Geek4414 (12000)
1271653	2012-05-03 03:47:00	Yep thats fine. The first thing you ideally want to get setup is the SIP Extension on the SPA3102 side of things. Have that able to make calls to your other softphones. Once you've done that, then worry about integrating your physical phone line in to it afterwards. Step by step. The Dialplan is basically irrelevant by and large, as thats what the SPA3102 will send through to Asterisk. So, if you match one of the settings then it'll pass the call and dialed number through to Asterisk which will then *make* the call as such. Use the dialplan from 2talk in their SPA2102 documentation at blog.2talk.co.nz :) It's NZ-specific! So, get the Ext setup in the SPA3102 and talking to your softphones and go from there!	Chilling_Silence (9)
1271654	2012-05-04 11:46:00	Yep thats fine. The first thing you ideally want to get setup is the SIP Extension on the SPA3102 side of things. Have that able to make calls to your other softphones. Once you've done that, then worry about integrating your physical phone line in to it afterwards. Step by step. The Dialplan is basically irrelevant by and large, as thats what the SPA3102 will send through to Asterisk. So, if you match one of the settings then it'll pass the call and dialed number through to Asterisk which will then *make* the call as such. Use the dialplan from 2talk in their SPA2102 documentation at blog.2talk.co.nz :) It's NZ-specific! So, get the Ext setup in the SPA3102 and talking to your softphones and go from there! Thanks for the link to the 2talk blog, I have plugged all that into the SPA3102. I now understand "a bit more" about all the parameters in the SPA3102. Progress from last night ... Anyway, finally decided to give 3CX another spin after having the issues of the network bridge going to sleep in the VM host. 3CX is installed on the Windows XP box, not as a VM and this is the box that I will leave running anyway, thought it might be simpler. Now what I have done now is ... Setup a Trunk call 10000 and have 3CX "log into" the SPA3102 PSTN line Linked this trunk to extension 101 (my softphone) Setup Extension 168 on the Line1 of the SPA3102 and have that "log into" 3CX (the analog phone attached to SPA3102) Now I can call between the extensions 101 < --- > 168 or other softphones etc Also can dial from 168 -> 101 although there is a funny dial tone and a bit of delay (silence) between placing the call and it ringing on 101, but it does get there. I can also now making outgoing (pstn) calls from my softphone (ext 101 or 555) by prefixing the number with a 9 which get stripped. When I call 168, it will also get diverted to Ext 101 (Android phone), and I can leave voice mail. I can call extension 123, put in my PIN and collect voice mail left by internal user or outside callers; can also manage and change all the voice mail prompts from there. Tested this on the LAN as well as over the internet, without the need to VPN in. So I have achieved about 70% of what I wanted so far, just need to ring from outside and make sure it will get patched to Ext 101 and people can leave voice mail! See this JPG, hope the upload works .... bayimg.com (link might not work in here, you may have to cut and paste the url)	Geek4414 (12000)
1271655	2012-05-04 12:23:00	Progress from tonight . . . After many days of hitting my head on the wall, finally got it working!!! Thanks goodness for details in the 3CX event log, it gave me some clues as to why the incoming calls not getting passed . In fact, I can see that the caller ID (pixelated) was in fact passed onto 3CX as it shows in the log but the call was not forwarded to my Extn 101 as expected, due to "authentication error" . . bayimg . com/laofiaadg . jpg" target="_blank">image . bayimg . com (cut and paste this link into another browser window, might not work if you just click on it) After googling a bit more, found a suggestion to add a "Line" which I couldn't find on the 3CX GUI, but I took a punt and added a VoIP Provider; added 101 as the Auth ID and set a password and hey presto, the call then get passed on successfully . Now incoming calls from POTS get forwarded to my SIP phone . It will go straight to voice mail if my softphone is off, or I can pick up the call if I have WiFi or 3G . The caller does not have to know I am away as they simply called my landline number . And I can place any calls via my landline from my SIPphone, including 0800 numbers even when I am out of the country . Last thing is to make sure I can work out the dial plan to block all toll calls, especially 0900, just to be on the safe side .	Geek4414 (12000)
1271656	2012-05-04 12:48:00	Awesome, glad to hear it! :)	Chilling_Silence (9)
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