Grandstream HandyTone 502
HandyTone 502 (HT-502) from Grandstream is an analog 2xFXS, 2xETH VoIP Gateway (SIP ATA adapter) with the integrated router.
Fully focused on open standards, these VoIP-gateways guarantee work with multiple systems, Softswitches and servers from various producers, supporting the SIP protocol. The HandyTone solutions have been tested and successfully used by many operators of IP-telephony (VoIP) around the world.
VoIP-Gateways from Grandstream Networks are successfully used by more than a dozen operators across the globe. VoIP-gateways produced by Grandstream Networks have been known in the U.S. telecommunications market as a reliable, easy to set up and use, low cost solutions for service providers and end users.
Grandstream Handytone 502
You could explore about the device on Grandstream's web page. We've got the unmentioned features that I actually like. HandyTone 502 has made a nice 1st impression and I'm looking forward to getting my rotary phones fully working on it.
Automatic Electric was an independant supplier of telephone equipment over the 20th century, ultimately being purchased under the patronage of GTE, and in direct competition with Bell empire through its existence. It is the 'partnership written' Wikipedia article tells tale more. The Northern Electric 302 and Automatic Electric Monophone 40 almost ready to duel. They planned to do one more search for VoIP analog telephony adapters that will accept rotary/pulse dialing, the AE40 works fine and will be a newest addition to the analog remix. Now let me tell you something. Everyone else report of a ATA that has probably been still in service and accepts rotary dialing, actually, a commenter considered UTStarcom adapters. Let me tell you something. Grandstream HT502"/I have a 502 coming from an ebay auction and will report back on how it works out.
Mostly, modern features in Asterisk 8 may drag me comfortable out stability that is version Release candidate three of version 0 was announced past working week. Updated module has been the last link needed to allow immediately placing and receiving calls thru Google Voice, it's neither standard SIP nor XMPPJingle, which we hope has always been yet to come. From the Google Voice web interface, you usually can specify Gmail Chat as one of your own phones to ring on a calls. Let me tell you something. This comment from the Asterisk code repository shows methods to make outbound Google Voice calls in one outline from the dialplan.
Previous fortnight they wrote about ways to send a XMPP message with Caller*ID data to one or more users when a call comes in to your Asterisk server. Even though, this is an elementary, oneway notification and has usually been dead simple to set up. So here is a question. What when we want to interact more fully with the Asterisk server over XMPP? Besides, please note that I am referring to administrative interaction, not 'calllevel' interaction. Asterisk 6 adds a Jabberget command to the dial plan, which will allow some interactive instant messaging within a call context.
It turns out that Asterisk Manager Interface posts an event for every XMPP 'packetboth' outgoing and 'incoming so' writing a Manager application to interface with XMPP usually was a proper technique to go. Writing a daemon in Perl to listen on the Manager interface for XMPP messages. AMI module accessible on CPAN.
With we get events because Events ‘on', set up the AMI connection. The user this script logs in as must be in manager. Now look. Observe in a list of buddies from Asterisk's jabber. Nevertheless, asterisk would accept commands.
Set up an ordinary event loop that filters out just Jabber events. Notice that mine looks like that.
You should take it into account. Validate the XMPP message. a dnc command, that adds a number to FreePBX blacklist; figure out if it came from a valid buddy. How does it work? As well, it's 'simplejust' add Asterisk XMPP user to your buddy list, and IM it your commands.
You could implement all the Manager command set in this way, even implementing a really clunky attendant console or ACD supervision tool. They see this being useful as a tool for elementary callcontrol tasks and unsophisticated administration tasks that imagine otherwise have to be performed at command straight line or web interface, that should be a bit much. Furthermore, near 2008 beginning, they published VoIP providers for home use, where they listed my favorite telephony service providers, a peculiar amount which we have now abandoned since they no longer exist, have probably been no longer cheap, or something better came along. We have my current choices.
Of course google Voice isn't virtually a provider but received several mentions above. Notice that gV shortly. Stay tuned likewise for a post on XLink, which is a valuable an integral element of my home VoIP setup for a couple years now. Asterisk has had a Jabber module for small amount of years, officially supported since the module enables 2 dialplan functions. Thence, jabberStatus, to get presence to, data and JabberSend send a message. Besides, i've been sticking with the long support release right now, there's more XMPP/Jabber functionality in Asterisk 6.
The 2 Jabber functions are actually useful in a customer service where agents have some desktop with XMPP maintenance. Presence function may be used in call routing for instance, the JabberSend as well as logic could be used for screen pops of caller facts, possibly bundled with external facts from a CRM package. As a consequence, I wanted to do it FreePBX way, which means not to mess around with context first-hand, since I am running FreePBX. You should take this seriously. There was probably no direct XMPP configuration module from within FreePBX so they have a 15 percent-FreePBX, 85 per cent-configfiles solution.
Remember, be sure res_jabber. Now pay attention please. Review /etc/asterisk/modules. Configure a XMPP account and set up details in /etc/asterisk/jabber. The config file is 'welldocumented' so we won't go through it step by step. There's a sample jabber, when you don't have it. That is interesting right? Asterisk source tar and a bit of info on the voip info wiki. Now regarding the aforementioned matter of fact. Be sure to add buddies you want to be able to IM with buddy= lines.
Restart Asterisk and use jabber show connected from the CLI to see that the user is now logged in. Now regarding the aforementioned matter of fact. Review log files and fix until it successfully connects. Now please pay attention. Replace label with the set up in jabber. XMPP connection. So, replace someuser@jabber. ID of a buddy you defined in jabber. SOMEUSERSTATUS with whatever you like. Notice that the ExecIn case statement which checks SOMEUSERSTATUS 6 sends a message to the buddy in the event he/she has been online. You could use three to IM the buddy in the event he/she has been in attainable or Chatty state. Did you hear of something like this before? Do the same for any various users you want to IM on an incoming phone calls.
A well-known matter of fact that is. Reload dialplan at CLI with extensions reload and after that pick up an extension and dial 1099 or whatever you set up, above. Considering the above said. This will trigger Jabber commands and in the event you have been online, you will get a popup stating that there's a calls from our own extension. Experiment and tweak to our liking. Now to integrate this in FreePBX, we're going to add 1099 to extensions list rung on a calls. On top of that, on my scheme we ring all extensions when an external call comes in. Keep reading. to ring all extensions and trigger the Jabber functions also, they just added 1099 to the Extension bottom List on my default ring group.
You usually can as well add the newest custom extension number to the Custom Extensions module in FreePBX, simply for tracking purposes, in case you want. At a flea market earlier in year, my mom picked up a Northern Electric 302 telephone, manufactured in August 1948, for She gave it to me all along a latest visit and they set out to make it work on my home Asterisk setup.
Remember, it pretty short is, it's functional. All in all, that has been to say, it gets a dial tone from my analog telephone adapter and they cdecision calls on it. Lots of information can be found on the web. ringer probably was really weak as we think my ATA solely supports nearly 2, perhaps five REN or 302 maybe needs about that much current to ring bells. Besides, rEN but probably this wasn't standardized until the 500 set. What to do about rotary dial? Considering the above said. Digium IAXy ATA, now discontinuedbut until then, they have a workaround. On top of that, some time ago, an acquaintance gave me an old enough Sharp organizer, the kind that stores a couple hundred titles and phone numbers and plays touch tones in our handset for speed dialing. However, he looked with success for it in a drawer and gave it to me as a joke. Reality that switch it to manual mode and I now have a dialer for my antique phone.You should take this seriously. There are usually dozens of resources on web for folks who want to fix up antique phones. Oldphoneguy has usually been a big resource for restoring and wiring up quite old phones to work on modern lines. Of course bell scheme Memorial, which they mentioned in an earlier blog post, has some good historical info. In the meantime there is some rustling in VoIP blogs about a modern 'Asterisk distribution on a CD' called PBX In A Flash. Ward Nerd Mundy Vittles blog is always man behind this distro. Of course asterisk@Home and after all Trixbox user, he got Trixbox tired limitations distribution and intended to produce his own. Asterisk hacks that have come from Nerd Vittles and we have the feeling this distro will be full of clever configurations and good feature additions.
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