Stack Exchange Network

Stack Exchange network consists of 175 Q&A communities including Stack Overflow, the largest, most trusted online community for developers to learn, share their knowledge, and build their careers.

Visit Stack Exchange
8

I'm not sure if you are asking for DR advice in your situation or a more permanent solution. However, for DR, I would HIGHLY recommend Telecom Recovery We use them and love their service. Our local provider has failover (either automatic or we can force manual) to send our DID 100 blocks to multiple 888 numbers at Telecom Recovery that are tied into a ...


6

VOIP. Voice over IP. It requires quite a few more things to be configured than plugging a phone into ethernet, but that puts you in the neighborhood.


4

Because that's how capitalism works. Seriously. I don't think there's any technical reason for doing this, other than "why give away what you could charge money for?". A possibly related question is "Why is DSL always download-biased?", but that seems to be equal parts capitalism and supply/demand rules.


4

Your remote user will need to register their IP phone to your Asterisk-based system, which means you will need to either allow non-local addresses to register, or provide a VPN or similar tunnel so the IP phone behaves as if it is local. Once either of these are accomplished, you simply create a SIP extension and password on the Asterisk server, configure ...


4

How about cellphones and a bucketload of SIM cards? OR. Depending on how dirty you want to get.. Have you got a fibre connection (dark, ideally) to the datacenter? Get a pair of Optical Add/Drop multiplexers. Put one in the DC, put one in the Office. In the datacentre, have Verizon supply you with ISDN over fibre. Plug it into the OADM. In the ...


3

Would you be looking for a workaround or a replacement for the faulty service? Replacement: If you can get an additional data line put in, you reserve it for telephony and you can run VoIP across that. This can be a line out to a dedicated provider that deals with telephony themselves and you can get good quality from that. There are "hosted telephony" ...


3

It's been a long time since I've fiddled with a Polycom, but once you find it's IP address, open it as a webpage in a browser, then guess at the admin username and password (might be admin:admin). There's probably some kind of configs in there. Or you could, y'know.. Ask Polycom Support.


3

Hardware: Asterisk will cope with very small hardware spec, there's a lot of examples and discussion at http://www.voip-info.org/wiki/view/Asterisk+hardware+recommendations on this. I've run it up on desktop PCs that have been retired from everyday use in the past, though only for playing with. VM: In theory, yes. In practice, you may find that depending ...


2

It's explained in the IP6000 manual. You're looking for pages 3-3 and 3-14.


2

Haave you looked at asterisk? http://www.digium.com/en/ you can use standard x86 hardware and purchase the cards to build your own. This will also give you flexibility in that you can go to voip if you chose.


2

You're limited by the number of analog lines you have connected. Each line can handle only one call at a time - so no matter what settings you change, your system won't have any higher external capacity. Note that if any call has an endpoint outside your organization - whether an inbound or outbound call - it needs a free line. You need a bunch of pieces - ...


2

Take a look at this link, it will help you. Pay attention to the last column. http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml However g729 is not royalty free from what I understand. I use g711, the quality is great and at 64k per call you would need 8 calls to fill your pipe. g729 is most effective and utilizes ...


2

(a) and (b) are more or less equivalent. The only difference between those two approaches would concern clients that don't support SRV records. By and large, all NNI communication should support SRV records. Some user agents might not support SRV. All DNS-based methods ((a) and (b)) incur a delay if one of the cluster members becomes unresponsive: DNS ...


1

Thank you for all of your help. Finally, I read somewhere that it may be helpful to record calls, so I started using the Monitor() application. Finally, I found that Asterisk wasn't listening for silence until after it played a recording first. I found this by realizing that Monitor() didn't start recording until something was first played by Asterisk, and ...


1

Taking the leap your in the US the actual regulations are for local number portability so if your new provider doesn't have a local presence you may not be able to transfer a number to them. From the FCC Website Keeping Your Telephone Number When Changing Service Providers Under the Federal Communications Commission's "local number portability" ...


1

Yes, if you look at the RFC at https://www.ietf.org/rfc/rfc3261.txt It explains exactly how to do this. 20.18 Error-Info The Error-Info header field provides a pointer to additional information about the error status response. SIP UACs have user interface capabilities ranging from pop-up windows and audio on PC softclients to audio-only on "...


1

1) can be done by changing "Asterisk Dial Options" in "Advanced features" tab. You need add S(1200) S(x): Hang up the call seconds after the called party has answered the call. 2) no, you can't in default elastic install. However you can request expert do module for such task( need add into sip_additional.conf variables and use that variables ...


1

Of course, you will not only need the second card but 4 more working POTS telephone lines. If you might want more lines than four times the number of suitable slots in the server, consider renting ISDN lines instead if available, there are cards available which are quad BRI so eight voice channels to a card (and even quad PRI cards which do a ludicrous ...


1

If by "calling" you mean from the PSTN, then you will have to add another card. It should be pretty much as easy as adding the card and using the manual to tell you how to get Asterisk to use it.


1

Yes you've got it right... a bare minimum setup would be an Edge server and a Standard edition server with all other roles collocated and using SQL express. A dedicate SQL server is obviously superior. Ideally, the environment's web facing services would be protect by a third server running Forefront TMG or another reverse proxy software. Our training ...


1

You can get these for many of the SoundPoint IP range of Polycom phones, but AFAIK, I don't believe you can get them for the conference phones (which I believe the 6000 is)


1

Get a sympathetic business nearby (same building?) to let you piggy-back off their phone system.


1

Take a high-fidelity CD recording of your favorite song. Record it using the cheapest microphone you can find. Encode the recording with a lousy 8-bit audio codec optimized for spoken words. Play the recording back through a cheap speaker (and wiggle the wires). If you listen to the CD and the chain above side-by-side you'll hear how badly mangled things ...


1

All DECT handsets available share a common profile - the Generic Access Profile. So you should be able to use any DECT handset paired with your base station to make or receive calls. Other profiles however, are optional and mostly proprietary, so things like caller ID display, automatic date/time settings, the ability to choose the outbound line / set the ...


1

You should switch to queues instead of ring-groups for this task, you can confirm calls in queues just make the first queue failover to the second and make the cell phones call in an order. 5555551111,1 5555551112,2 More on queues module http://wiki.freepbx.org/display/FPG/Queues+Module+User+Guide


1

The number of lines are the number of active calls on that specific phone extension, depending on the business you'll want to stick them in a queue like recommended before and distribute them to different people/departments. In reality except for a receptionist, the use of more than two lines is usually worthless, and I'd usually stick them behind a queue (...


1

If you want to do SIP use: - FreeSwitch - Asterisk - Yate If you want to do H.323 use: - GnuGK Note: SIP still do not offer H.239 solutions with none of those. So after learning from faults i moved to GnuGK, which does all in H.239 (i always wonder why SIP was required).


1

(I am actually using Sipnet and Zadarma on my FreeSWITCH server) Sipnet.ru allows you to forward all incoming calls to a SIP URI. Also it allows creating sub-accounts. So, you create as many sub-accounts as you need, and set the forwarding of all calls to some unique URI, like vasya-pupkin@yourdomain.com Then in your FreeSWITCH configuration, you create a ...


1

VoIP servers (particularly Asterisk) are pretty notorious for having security holes - the degree of risk they pose being dependent on what they have access to (This guy has made a hobby of finding issues with VoIP servers, with Asterisk taking the bulk of the heat). Accordingly having a separate VoIP network (or at least a separate vLAN) with limited access ...


1

The simple answer is that your map will work: Your gateway router/firewall needs to know enough to send the "call center" traffic out to the XO T1's gateway as opposed to the Cox cable gateway, but that can be done with simple rules/static routes. Regarding routers and other terminating hardware for the XO connection, that's between you and XO -- They should ...


Only top voted, non community-wiki answers of a minimum length are eligible