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21

Yes. Being in a position currently where the powers that be bought a Panasonic PBX. There's a bunch of features that aren't there (or rather, are individually licensed). Such as MeetMe (conference calling), or hot-desking. There's also the problem that all the phones are effectively hard-wired to their ports. They had to run new cables for the phones, ...


18

In this day and age you cannot directly compare a soft switch to a PBX. A soft switch, or software that connects phone lines with one another, is at the core of all modern PBXs. Asterisk is not a PBX but it can be configured to function as one. Also most of the functions provided by an PBX can also typically be performed by a class 5 soft switch. Perhaps an ...


14

It's a simple trade-off of cost vs. risk. If you believe (or have anecdotal evidence or reviews) that Cisco gear would be rock solid, then it might be worth spending the extra money. Phone systems can be pretty central to a business' operation. For some companies, saving (for example) $100,000 on a phone system is peanuts compared to the cost of 20 minutes ...


12

All the answers to your questions are "it depends". If you use a PBX system like Asterisk where the audio data is actually handled by the server computers you'll have much steeper CPU and I/O demands on the server computer (along with finicky reliance on timing-- something that virtual machines don't necessarily do a great job with). If you a PBX system like ...


9

Virtualizing a PBX is a challenge, due to one main aspect: There is no guaranteed scheduling of your PBX VM and the general scheduling behavior could introduce jitter. That being said you also have to think about how your line-cards (if you need S0 to some other PBX etc.) need to be presented to the VM and if things like vMotion and HA make sense. There are ...


9

CPU consumption should be minimal, for the most part it resides in the network libraries / ip-stack. murmur doesn't do decode-multiplex-reencode. The audio-streams are forwarded/copied to the appropriate clients verbatim. RAM working set are limited to input/output buffers for the clients (which would be small since one of the project goals is optimizing ...


9

You can find lots of good royalty-free music at http://incompetech.com/m/c/royalty-free/. He's got lots of original content that is suitable for music on hold, and we use several songs available there. Coincidentally, I found that site by Googling for "royalty free music".


9

A good rule of thumb comes from old telephony .. an phone T1 line 24 voice channels and a control channel. This equates to roughly 64Kb/s per phone line. This rule works (roughly) for uncompressed VoIP. There are various codecs that do compression .. the G729 should roughly triple that, so get 72 voice channels on a T1. If you expect all 55 phones to be ...


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 ...


8

IPX? :-) Doing a google search on the phrase you're asking about seems to indicate that you can't send calls over this VoIP link if it originates on POTS. If someone calls into your HQ office from a POTS line, and you then route it to one of your branch offices over your provider's VoIP link, you're in breach of your contract, and they're probably in ...


7

I have come across the same problem and it's not just limited to asterisk. In the end we came up with something that worked well for us. We called it a SIP Loopback. Basically we signed up for an ALT sip provider (flowroute.com) and setup a script that calls out via primary SIP provider to the phone number setup with our ALT provider three times an ...


7

TCP retransmissions are usually due to network congestion. Look for a large number of broadcast packets at the time the issue occurs. If the percentage of broadcast traffic in your capture is above about 3% of the total traffic captured, then you definitely have congestion. Look for both physical layer (ARP) and network layer (name resolution) broadcasts on ...


7

Your networking issues will increase exponentially. You'll have to troubleshoot issues with latency from various locations, you'll have worse packet jitter, etc. I've had nothing but heartache with SIP to remote sites/servers.


7

ServerFault doesn't really do shopping advice - you need to consult with vendors and decide what system works best for you. Some points to consider: Do you want a fully VOIP system (IP within the office, your PBX connects via VOIP to a provider that connects you to the PSTN), or a hybrid system (IP within the office, and your PBX connects to the PSTN ...


6

let others do it. if you want to get a reasonable answer i suggest you add [much] more details - eg: what voip protocol does your service provider handle [probably sip, but it's worth checking] are your internal users going to use soft or hard-phones what features should the telephone system offer - just calling in/out? call groups? auditing/recording? ...


6

You really should hire a reputable network professional to help you. You don't need a second router, although you may need a better router than the one you currently have. I'd say that you're asking good questions but lack a lot of basic networking know-how, and simply need a seasoned hand to do this for you. In the contract, include some hours for knowledge ...


6

Your setup is suboptimal -- I'd even go so far as to say that the setup you described is actively hostile to VoIP communications. A few key items: 1. You're sending video and audio both ways over the RDP session. 2. You're going over a wireless network. While this often works fine it's generally not a great idea, especially on common WiFi (not dedicated ...


6

Nothing from hard experience, but I would take it as a given that PoE switches draw more current than their regular cousins. Cisco has some numbers for Catalyst 65xx series equipment. To summarize a 48-port 10/100/1000 board dissipates 443 BTU/Hr and the PoE version of that board puts out 518 BTU/Hr (+17%). To be safe I'd inflate the heat load by ...


6

For years we played the CDs of a band owned by a friend of mine (written and produced), it was jazzy with Latin flavors. He allowed this to happen, so no worry about paying him royalties. An idea would be to talk to a somewhat good local band(s) to see if they'd be willing to do the same thing. I'm not aware of online subscriptions... I certainly wouldn't ...


6

I've had great luck with Polycom phones. I've used the IP450 extensively in conjunction with our asterisk server, and have never had a single problem. I've also used the IP4000 conference phone in the past, and it's performed very well. If you're looking for something a bit cheaper, try the Linksys (I guess it's Cisco now officially) SPA-941. I have a dozen ...


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 ...


6

You can use tools like WANEM to simulate packet loss. It's a Live-CD, so you can put it on a system between your server & clients.


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.


5

Phone numbers and "SIP URIs like me@domain.com" are different kinds of URIs. A phone number can be represented as a URI like this: tel:+12125551212 whereas a "SIP URI like me@domain.com" can be represented as a URI like this: sip:me@domain.com Many SIP user agents don't actually use tel: URIs. Instead they just take the number that the user dialed and ...


5

It works but there are applications that depend on an accurate clock that need some special attention, for example conference bridging. A good test is to install asterisk + zaptel and compose a test dialplan for a conference bridge. Connect to the bridge with two phones and listen to the audio.


5

We're using Grandstream products - the speakerphone is OK, they're dirt cheap, and my users have been pretty pleased with them. You can also get excellent Cisco phones under $300. If you haven't looked there yet I would suggest checking out voipsupply.com -- their prices are decent and the user reviews are usually helpful.


5

A lot of this depends on the bandwidth that your phone system is going to use - if it's a call center with hundreds of people on the phone simultaneously you'll have a higher need for a separate network than an office of hundreds who just want phones on their desks but aren't heavy users. A conservative (as in, "way more than is really used") estimate is ...


5

Microsoft Office Communication Server 2007 R2 is an excellent product. It has all the features you've mentioned and more should you need them (like integrating with hardware PBX's and what not). In my organisation we have 300 employees across 2 geographically distant sites so we had a Standard Edition OCS server at each site. The only downside is there is ...


5

Skype



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