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<body class='hmmessage'><div dir='ltr'><font style="" color="#000000" face="Tahoma,sans-serif">An ATA is an analog telephone adapter. I use the Granstream HT503 here and it works well. For years I had been using the Linksys (Cisco) SPA-3000 but it constantly gave me problems. They have both FXO and FXS ports which are more confusing terms. In simple terms Foreign Exchange Subscriber (FXS) s the output to the local phones and Foreign Exchange Office (FXO) is the output to the central office. In many cases you would just be using the FXS port to local analog phones. The ATA supplies dial tone to the local phones and provides the A/D and D/A functions. This can be likened to the USB FOB connecting your radio in Allstar.<br><br>The ATA is not even necessary if you use a digital phone to begin with. While they are more expensive there are some that are in the $50 area. <br><br>The ATA or digital phone connects via Ethernet. They are just another IP address on your LAN. Generally you configure the ATA or phone with the IP address of the Asterisk server and then setup sip and extensions in Asterisk. The phones can call each other or if you have an FXO port setup out to a central office. You can also setup a VOIP provider to connect to over the Internet.<br><br>Doing this manually is probably more than most want to take on. You have to know the Asterisk language and method of doing things. That is why I suggested a standalone server like Incredible PBX which would be much easier to setup and use. A connection between the Asterisk Allstar server and the PBX server would be via IAX over the local LAN. The other advantage of using one of the new images is that you are using one of the latest or later versions of Asterisk. Asterisk Allstar is using version 1.4.23. The 1.4 version release was 12/2006, almost 10 years ago. Many of the current examples might not work with the older code without modification. <br id="FontBreak"></font><br><font style="" face="Tahoma,sans-serif">The setup for a specific device would be unique to that device and usually you can find manufacturers examples for sip stanzas. This is in sip.conf. Here is one for a Granstream Digital phone I have on my PBX -<br><br>[grandstream409]<br>type=friend<br>host=dynamic<br>context=default ; Where to start in the dialplan when this phone calls<br>callerid="Doug Crompton" <407> ; Full caller ID, to override the phones config<br>nat=no ; there is not NAT between phone and Asterisk<br>canreinvite=no ; allow RTP voice traffic to bypass Asterisk<br>username=grandstream409<br>regexten=409<br>secret=<ADD PW><br>dtmfmode=rfc2833<br>;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time<br> ; from the phone to asterisk<br> ; (1 for the explicit peer, 1 for the explicit user,<br> ; remember that a friend equals 1 peer and 1 user in<br> ; memory)<br>mailbox=405 ; mailbox 1234 in voicemail context "default"<br>progressinband=yes<br>disallow=all ; need to disallow=all before we can use allow=<br>allow=ulaw ; Note: In user sections the order of codecs<br> ; listed with allow= does NOT matter!<br><br>And for the Grandstream ATA device (FXS)<br><br>;Grandstream HT503<br>[ht503fxs]<br>type=friend<br>username=ht503fxs<br>secret=<ADD PW><br>host=dynamic<br>nat=no<br>dtmf=rfc2833<br>port=5060<br>canreinvite=no<br>disallow=all<br>allow=alaw<br>allow=ulaw<br>qualify=yes<br>context=default<br>regexten=405<br>callerid="Doug Crompton" <405><br>mailbox=405,406<br><br>So this particular Grandstream phone is on extension 409 and my analog phones are on extension 405. These are just examples and would probably be significantly different depending on your hardware and configuration. You would also need to setup extensions.conf to service the sip and/or iax ports. <br><br>So that is a little background. Getting into the phone end of things is really a little off topic for this list. There are lots of Asterisk PBX forums. Here are a few links -<br><br><a href="http://incrediblepbx.com/" target="_blank">http://incrediblepbx.com/</a><br><br><a href="http://nerdvittles.com/?tag=incrediblepbx" target="_blank">http://nerdvittles.com/?tag=incrediblepbx</a><br><br><a href="http://nerdvittles.com/?p=14787" target="_blank">http://nerdvittles.com/?p=14787</a><br><br><a href="http://nerdvittles.dreamhosters.com/pbxinaflash/downloads/" target="_blank">http://nerdvittles.dreamhosters.com/pbxinaflash/downloads/</a><br><br><a href="http://pbxinaflash.com/community/index.php" target="_blank">http://pbxinaflash.com/community/index.php</a><br> <br><br></font><b><font style="font-size:16pt;" size="4">73 Doug</font><font style="font-size:16pt;" size="4"><br></font><font style="font-size:16pt;" size="4">WA3DSP</font><font style="font-size:16pt;" size="4"><br></font><font style="font-size:16pt;" size="4">http://www.crompton.com/hamradio</font></b><font style="font-size:16pt;" size="4"><br></font><br><br><div>> From: kk6ecm@gmail.com<br>> To: arm-allstar@hamvoip.org<br>> Date: Wed, 16 Sep 2015 21:00:29 -0700<br>> Subject: Re: [arm-allstar] adding ATA's<br>> <br>> Newbie here... I don't recognize the acronym "ATA." I'm guessing it means<br>> something like Analog Telephone Adapter?<br>> <br>> Spent my whole career putting up with acronyms... each discipline seemed to<br>> have their own set of definitions for the same acronym (sigh) so never have<br>> been too fond of them :-) <br>> <br>> http://www.brinkmanonline.com/humor/church/wc.html<br>> <br>> So... definition for ATA, please.<br>> <br>> Thanks,<br>> Bob<br>> kk6ecm<br>> <br>> -----Original Message-----<br>> From: arm-allstar-bounces@hamvoip.org<br>> [mailto:arm-allstar-bounces@hamvoip.org] On Behalf Of Ken Page<br>> Sent: Wednesday, September 16, 2015 8:36 PM<br>> To: arm-allstar@hamvoip.org<br>> Subject: Re: [arm-allstar] adding ATA's<br>> <br>> Hi Doug & Co.,<br>> <br>> RE: adding ATA's to our PI2.<br>> <br>> We may add additional PI2's later more so for load sharing should we <br>> find they can't handle multiple things at once.<br>> <br>> However for now we'd like to try it on the one we have set up now.<br>> <br>> Also back when we ran Gentoo Linux for our All Star the all seeing all <br>> knowing Duuude had things set up for us so we could dial in from our <br>> ATA's enter a PIN and gain access to the All_Star network from the <br>> phones locally without having to use a radio.<br>> This would be a nice feature to have again so that in it's self might be <br>> enough to want it all on the same device instead of trying to link two <br>> together.<br>> <br>> Soon as I get a chance I'll start Googling how it's done but if anyone <br>> wants to throw up a few suggestions or examples it would speed things up <br>> a lot.<br>> <br>> Our last system was some years ago and I think we were running Asterisk <br>> 1.2.x originally from memory.<br>> But it's been a lot of years since I have tinkered with such things.<br>> <br>> <br>> Regards,<br>> Ken<br>> .-.-.<br>> <br>> Message: 5<br>> Date: Wed, 16 Sep 2015 09:04:34 -0400<br>> From: Doug Crompton <doug@crompton.com><br>> To: ARM Allstar <arm-allstar@hamvoip.org><br>> Subject: Re: [arm-allstar] Some Hints to add ATA's needed please?<br>> Message-ID: <BLU171-W857E69E9145DDB87A74E60BA5B0@phx.gbl><br>> Content-Type: text/plain; charset="iso-8859-1"<br>> <br>> While this can be done I would advise not mixing on the same system as <br>> Allstar. As cheap as these boards are why not run a separate board to do <br>> that. There are many ready to go images out there like incredible PBX <br>> etc. They use much later versions of Asterisk and have much more PBX <br>> capability and run in a graphical environment on the BBB or RPi. Later <br>> on you could connect the two via your LAN just as well as having them on <br>> the same server<br>> <br>> That being said if you really want to use the Asterisk Allstar image <br>> also as a PBX it can be done by adding a sip account for each individual <br>> phone with a corresponding extension stanza. It would have to be done <br>> manually and it would require a lot of experimentation which would <br>> require restarts that would disrupt the radio side of things. You also <br>> would have to follow the Asterisk 1.4 or earlier command set.<br>> 73 Doug<br>> WA3DSP<br>> http://www.crompton.com/hamradio<br>> <br>> <br>> _______________________________________________<br>> <br>> arm-allstar mailing list<br>> arm-allstar@hamvoip.org<br>> http://lists.hamvoip.org/cgi-bin/mailman/listinfo/arm-allstar<br>> <br>> Visit the BBB and RPi2 web page - http://hamvoip.org<br>> <br>> _______________________________________________<br>> <br>> arm-allstar mailing list<br>> arm-allstar@hamvoip.org<br>> http://lists.hamvoip.org/cgi-bin/mailman/listinfo/arm-allstar<br>> <br>> Visit the BBB and RPi2 web page - http://hamvoip.org<br>> <br></div> </div></body>
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