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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.


6

Well that looks like a PBX for a phone system to me. Above it to the right is a punch down block where, presumably, they have POTS lines coming in to "feed" the PBX.


6

Definitely research is going to be your best friend. You'll soon realize that a VLAN capable infrastructure is key for your VoIP success (meaning: keeping your VoIP traffic logically separated from your 'normal' hosts). Also, make sure your network infrastructure switches/routers are going to be up to specs for VoIP: Full-duplex, QoS Capable, and free of ...


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

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

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 ...


4

You don't appear to have considered the simple option: SMSes. Saves all the hassle with text-to-speech and Asterisk insanity.


4

If you have the bandwith (roughly 100 kbps per call), G711 should give you best call quality (and least CPU usage in case Asterisk is transcoding). Otherwise, G729 is a good choice. These two should be supported by most hard- and softphones. For more exotic codecs, check what your phones support.


3

One of the most common misconceptions about voice is that it's easy and you can just whack it on a network like all the other traffic. That approach can work, but you really need to do some prep work. First of all, make sure your network is up to it. There are plenty of network analysers out there that will run over your LAN and report packet loss, latency ...


3

We once did it ourselves with Asterisk. We're small, so eventually we decided to go with Trixbox. They have a very sucky billing department, but their support is good (sometimes very slow do) and it just works. As far as terms go, you pay for the licenses and support, since it's a whole PBX, not just a conferencing solution. You can use either land lines ...


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

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 ...


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.


2

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


2

There is a ton of information here: http://www.voip-info.org/ It's mostly related to the open source solutions so it may not be what you're looking fo but there is still a good amount of info. Especially on the phones themselves.


2

I have heard that the Digium cards are ideal for interfacing Asterisk with the UK Analogue PSTN. Digium was started by one of the initial developer of Asterisk, so compatibility is there by design. Digium AEX410 looks like it would suit your needs... http://www.digium.com/en/products/analog/


2

I would suggest that you use g729. You will need to buy some licenses from Digium ($10 each, only need to buy enough to support expected concurrent calls). It is money well spent. Cheers


2

There are a number of variables to take into account when looking at transitioning to a VoIP system. The first thing you have to decide is what type of system you are going to want to implement. When it comes down to it there are basically four types of office telecom systems (all terms mine). Traditional - This is what you would think of the classic PBX. ...


2

I vaguely remember someone suggesting that with your own asterisk server, you could add a gateway to your network which would route skype call to your standard SIP IP phone. The thought of being able to answer skype calls using your normal voip phone is quite appealing, but I haven't had the time to look into it any further. Plus, if you have both a SIP IP ...


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

(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 ...


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

The answer is, it depends. The modem you are wanting to use needs to be a VoiceModem to record the calls. If you are not recording and just detecting off hook duration any modem could work. If you want to do recording and playing sounds back to the caller you will need a modem that supports full duplex voice audio.


2

Capturing CDR/SMDR data from the serial port could seem like a simple job but it isn't. The serial communication is very sensitive and it is not uncommon to lose data. For example a GSM phone that is less than 2 meters from the cable can cause big damage to the communication. You will still receive data but it will contain errors and you need a very smart ...


2

Hosting internally Pros: More control over costs and technology Cons: Increased cost for line usage to the PSTN. Outsourcing: Pros: Can Scale on demand, less management, more features (ie transcription) Cons: Pay per swallow. I've used: Freeconference.com if all you need is audio. Max size 150! Hey Otto has a certain cool factor to it, but it's $.09 per ...


2

We use NetSuite and have a Cisco Phone system. We do not do exactly what you are looking for as our integration piece is from a Customer Service stand point with Cisco's Unified Contact Center Express. Our process is basically a customer calls in to our Support line, is asked for their Customer Number and is then put into the Queue for the next agent. From ...


2

take a look at snom. i've read lot of good reviews of it. i bought recently two snom 300s but it's too early to comment on them. avoid siements - i used s675 ip and s685 ip - with few handsets associated with base stations they hang every few weeks.


2

By voice, do you mean 1.) you're using a MUX to split channels out of the current T1 for voice, or 2.) you've got VOIP as part of your data traffic? I don't think there would be any difference whether your link was 1 T1 or a MPPP link, as long as all of your MUX and termination equipment supported MPPP. You'd have to confirm that it would work though, so ...



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