The aid two or more services of VOIP

Hi,
Any person does know if to the 3102) aids two of the BATH 3000 (or more registries of the VOIP supplier on watch that is similar to SPA-1001 ? virtual Port of the services ? (VSP). According to the Web site of Sipura, the VSP allows that one is placed with two independent services of VoIP. Then the entrance calls can be alerted that using a distinguishing ring and calls of exit they are supported using plans of the dial. The SPA-1001 also supports the communication etc. between the two calls.

Hay a certain thing similar to VSP in the SPA3000 or the SPA-3102? (that is to say, a device similar to SPA-1001 with FXO).

Grazias.



Quotation:

Fixed originally close sandun Hi, Any person does know if to the 3102) aids two of the BATH 3000 (or more registries of the VOIP supplier on watch that is similar to SPA-1001

The answer is no he no. If their necessities are exactly two suppliers of VoIP, you are IMHO better extinguished with the SPA-1001.

The SPA3000 (and probably also the 3102, but I have only the 3000) has only ONE ? grooves of the registry ? for ? main line 1 ? line (and one 2da grooves of the registry that can be only used in the side of the PSTN of the adapter for dialin to the characteristics of ? entered ? of dialout).

Now SPA-3000 ? considers several suppliers of exit via the 4 the grooves of the entrance ?, and ? extended plan of the dial ?. So if you as soon as you wish the suppliers in who you can choose mark outside towards the time (via modifying his ? plan of the dial for particular requirements ?), SPA-3000/3102 is great.

And you can even use varies ? trampea expedition ? to obtain several suppliers of entrance. For example, I invented this ? trick ?, so I know that it works (if a little TOOTS to setup):
Wiki runner of the SIP: Calls of entrance directly to its LinkSys or Sipura


Thanks to respond.

It is for the members of the family of the non-techie (who wishes calling-ins from 2 diverse suppliers), so I think that I will obtain BATH 1001 for them. They can still keep the local line of telco/PSTN (on a separated telephone) if they have pleasure. (although I think that they will obtain drawees of him once hooked in voip).

Thanks for the connection to the entrance of Wiki. I am also new to VOIP (that uses a Atcom at the moment 168V TIES - hoping to obtain to the 3000 or the 3102). Therefore, watched as it can be that it can use that example with dyndns.org (that is what I am using at the moment for dns dynamic).

A certain one what asks without relation. I soiled around with using a PBX - (Axon) and tested a PSTN of the microtelephone> IT TIES> PBX> SipProvider-> of the call (that is to say. say aloud to my moving body). I notice that a slight one delays. It is this latency due to the Axon.

Grazias.


Quotation:

Fixed originally close sandun It is for the members of the family of the non-techie (who wishes calling-ins from 2 diverse suppliers), so I think that I will obtain BATH 1001 for them. They can still keep the local line of telco/PSTN (on a separated telephone) if they have pleasure. (although I think that they will obtain drawees of him once hooked in voip).

All the LinkSys/Sipura adapters of the IMHO is TOOTS so that no-techy setup and maintains. Nevertheless, if the systems techy he stops above for a family (and it eliminates errors who of her if/when there are problems), they ? are used ? like a normal telephone line simply (despite with options possibly more than they mark). If you really wish something easy setup, obtains a PhoneGnome. Nevertheless, the complexity of the disposition even exists there, he is joust that the handles of PhoneGnome.com that complexity for you ?

FWIW: I make a very flexible SPA3000/advanced setup in the country. And I know that I am the unique one in my family who can make the heads or the tails of real configuration SPA3000. And I am also the unique one who at the moment understands the options that mark true of ? supposition ? (such as the capacity to directly mark to the directions of free VoIP ? for ? with ? running of the SIP ?). Nevertheless, that really does not matter to the rest of the family, as far as them ? line 2 ? (the exit of my SPA3000, according to the sight by our telephones without threads), is right other more ? telephone line ? with some ? options ? than (telco/POTS) ? main line 1 ? has. For example, line 2 allows limitless ? LD ? that marks in the E.E.U.U. And ? also aligns 2 ? has ? caller identification ? (and it equals the identification of caller modified for particular requirements for some specific numbers) when people call pulg. But of another way, ? seems/acts one normal telephone line ? to the other members of the family. I am only which it must worry about the complex details of the disposition, and also the unique one that uses the options ? outposts ? that setup (for example the option to make VoIP free to VoIP calls, bypassing our supplier of VoIP). But to the family, ? can nevertheless use easily ? telephone line, as soon as like any other telephone line (I setup once the adapter for the house).

Quotation:

Fixed originally close sandun Thanks for the connection to the entrance of Wiki. I am also new to VOIP (that uses a Atcom at the moment 168V TIES - hoping to obtain to the 3000 or the 3102). Therefore, watched as it can be that it can use that example with dyndns.org (that is what I am using at the moment for dns dynamic).

You must be able to do that. I use that same trick with the free service of no-ip.com (another dynamic company of the DNS). And also I have heard information of the people who used that trick with dyndns.org also.

And the really pleasant thing, is that the trick of the expedition can be used in addition his ? registry ? in the SPA3000. This does not prohibit ? him normal registry ? + no number of suppliers front you ? ? to its dynamic DNS (possibly indirectly via a runner of the SIP ? alias ?, reads notes in the trick on the difference between DNS ? SRV ? registers carefully that certain suppliers of VoIP/necessity of the equipment, and the DNS ? To ? registers that most of the dynamic source of the companies of the DNS). Essentially they do not limit to him so how many supplying/DIDs you must come inside with this trick. Nevertheless, it can still be TOOTS to be able that trick to work initially to the right (you must obtain all the correct details, including correctly setting-up your rebajadora for the traffic ? not asked for ? of VoIP that an incoming call WITHOUT the registry appears to be).

BTW: That does not forget that the SPA3000 is very good in having options to call of exit, that you can choose in the time of the dial (if you make his ? plan of the dial ? form to take advantage of this flexibility). I personally have not only my main supplier of VoIP, but several other places that I can call (cualquieres places including accessible via ? running of the SIP ? for free), simply marking sequences that mark of the custom setup in my plan of the dial. For example, my plan of the dial setup to normally dial 11 numbers of the digit via my main supplier of VoIP. But frame a 1 additional one opposite, instead of another shipment the call to a diverse account (of reserve) of VoIP. And 11 in front of the number, sends to my 2da reserve account. And if instead of another frame a sequence beginning with the *-key, ? leaves via ? the free service of the runner of the SIP (and that one is right some of the options that I have put in my plan of dial SPA3000). Whereas they are not the defect (you must know how to use the multiple accounts of VoIP in SPA-3000), the options that call of exit (with ? plan correctly modified for particular requirements of the dial ?) are almost as flexible so many systems of the PBX offer!

Quotation:

Fixed originally close sandun A certain one what asks without relation. I soiled around with using a PBX - (Axon) and tested a PSTN of the microtelephone> IT TIES> PBX> SipProvider-> of the call (that is to say. say aloud to my moving body). I notice that a slight one delays. It is this latency due to the Axon.

In most of the cases, the latency/delays in a local PBX is generally minimum (a fraction so small second, as far as is not sensible by oído/la mind human). The problem comes on when you go you release distances ? of the site ?.

IMHO that the most probable source of his delays, that is the traffic of the Internet between its PBX and its supplier of the SIP. He is not infrequent for the latency of the Internet/delays, between an adapter of VoIP and a supplier of VoIP, to be 1/3 greater than of second all by itself. Whereas 1/3 of a second are not even enormous, it is than enough more so that the human ear notices a slight pause. And if the equipment of those of the telephone in the OTHER END OF the CALL again escaped ANYONE of its voice to you, ? even so short ? delays will sound like ? echo to him ?. ? is some trampea him ? can make to form ? echo protects ? and such, but in the end, this one is one of the limits of the transmission delays caused sending its voice on the general Internet (that is to say. a VoIP problem can reduce to the minimum by the appropriate configuration, but an intrinsic problem of ? latency ? in the Internet itself).