RECOMMENDATIONS New Deployment Plan for a non-profit

Robert-BCC

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Hi,

I am an IT volunteer for a Breast Cancer support organization with ~10 employees and 20 phones that will be moving to a new location in late August. They’re currently paying AT&T over $600 / month for 1600 minutes of monthly usage. Currently, a T1 connects to an ancient PBX which has an IVR and services 10 DIDs, including one toll free number and two fax machines. They say they'd have no more than six simultaneous calls, max. This solution has been in place for a decade, is rock solid, and completely unsupported. For this organization, voice is more important than data.

We are ordering two ISP connections and two POTS lines. A voice VLAN will carry traffic exclusively to the slower ISP connection (5Mbps down / 1 Mbps up). Thankfully, the carrier can enact AnnexM to make the link more symmetric. They guess that will get me roughly 3 down / 2 up. I'm planning to have all numbers ported to Vitelity.com

Although I have a background in networking and telecom, and some linux skills, I’ve never setup or managed a PBX. To begin learning, I’ve setup my own PIAF (Purple) but I really need to figure out more about the basics of this deployment. Here are my initial questions:

- Will the bandwidth be sufficient using default codecs?
- Should I consider doing this entirely in the cloud (aka rentpbx.com)
- If I do this deployment on-site, why would I want/need a line card?
- I'm thinking of just running the two POTS lines directly to the two fax machines, and bypassing the PIAF server entirely. Is that smart?
- If you live in the San Jose area and would like a free lunch this week, I'd love to pick your brain!

Thanks,

Robert
 

rossiv

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Welcome to the forums, Robert-BCC !
- Will the bandwidth be sufficient using default codecs?
ULAW is generally the default codec. Uses ~64kbps per channel, so roughly 3x2 should work if it's reliable. That really depends on how far you are from the CO and a zillion other things with DSL.

- Should I consider doing this entirely in the cloud (aka rentpbx.com)
Personally, I'd refrain from doing that just because of your bandwidth.

If I do this deployment on-site, why would I want/need a line card?
I'm thinking of just running the two POTS lines directly to the two fax machines, and bypassing the PIAF server entirely. Is that smart?
Lumping these two together. No line card needed from what I can tell unless you want analog extensions from PIAF. Just go directly to the fax machines.

If you live in the San Jose area and would like a free lunch this week, I'd love to pick your brain!
I'm in Atlanta myself, but I am more than willing to have my brain picked, and I'm sure the other gurus here would be willing to contribute as well.
 

atsak

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You'll be fine; consider using one of the lines as a lifeline out if there's a problem; you can connect both the fax and the PBX to it using something simple like an Obi 110 for the PBX. Then calls can still go out if you have an internet outage (be sure to install a recursive DNS server on the PBX and make it the primary DNS resolver). Certainly not necessary; in many areas DSL is getting much more reliable (only outages seem to be for maintenance at night around here for the last 18 months or so).
 

Robert-BCC

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First, thanks for the rapid, welcoming, and thoughtful responses. I'd like to expand on the future of BCC's network connectivity. Although the VoIP line will be roughly 3Mbps down / 2Mbps up, a separate Comcast Deluxe 50 connection exclusively for non-voice traffic is marketed as having 50/10. We'll also have some fairly studly donated switches and firewalls in place (QoS, PoE, complex ACL support, SSL VPN termination, etc.) On the rentpbx discussion:
Personally, I'd refrain from doing that just because of your bandwidth.
Excluding peer to peer calls, wouldn't the required bandwidth be the same if the PBX was inside or outside the org? And if we have 6 calls X 64kbps [384kbps] max traffic, wouldn’t our dedicated 3/2 be sufficient for all voice traffic headed to the cloud PBX? I conceptually like the cloud PBX idea for failover (if one ISP connection dies) although the security aspects of a server sitting on the net are disconcerting. And then I’m not sure how we’d implement atsak’s suggestion:
You'll be fine; consider using one of the lines as a lifeline out if there's a problem; you can connect both the fax and the PBX to it using something simple like an Obi 110 for the PBX. Then calls can still go out if you have an internet outage.
I like this idea a lot and I can setup the recursive name server. But...
- How would I service the second fax machine? (there's no need for them to be in use at the same time)
- How would the failover to the Obi 110 work? Our Internet connection is down, a client places a call to the toll free number, how would Vitelity detect the failure and know to route the call to the Obi 110's POTS interface?
- Presumably I would program the Obi 110 to route all incoming calls the front desk phone? (I had them buy a bunch of Grandstreams btw). I have more questions, but that’s enough for tonight!

Robert
 

jeff.h

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If you have a quality router that can do QoS you might actually consider putting the voice on the Comcast connection or even better yet, get a router than can do dual WAN with failover if you want to keep two connections. There are a few that come to mind, the ASUS wifi routers and the Cisco small business routers can do this.

PIAF without Incredible PBX does work nicely as an inbound fax server as well. I have a few in place, it can auto detect faxes and send them out as PDFs in email.
 

atsak

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I like this idea a lot and I can setup the recursive name server. But...
- How would I service the second fax machine? (there's no need for them to be in use at the same time)
- How would the failover to the Obi 110 work? Our Internet connection is down, a client places a call to the toll free number, how would Vitelity detect the failure and know to route the call to the Obi 110's POTS interface?
- Presumably I would program the Obi 110 to route all incoming calls the front desk phone? (I had them buy a bunch of Grandstreams btw). I have more questions, but that’s enough for tonight!

Robert


1. Just split the regular phone line or output the line from the Obi and set the fax to not answer.
2. I don't know if Vitelity offers failover service, but if they do it's just a setting in their portal. There are other carriers that definately do if tehy don't.
3. You program the obi to route the calls to the PBX, it can go to the front desk if you want or you can send the calls in the same "route" as the SIP calls if you want; whatever works.

I was in fact thinking more about outbound for you than inbound but nothing stops you from using it inbound. For companies that live and die by their phones I usually suggest a line or two setup this way, for the once a year there's a problem of some sort.
 
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It sounds like your ISP connection is going to be DSL. One thing that seems endemic to asymmetric DSL connections is something called buffer bloat (http://en.wikipedia.org/wiki/Bufferbloat). If your ISP's setup is susceptible to buffer bloat, it will absolutely destroy your SIP call voice quality when someone is generating upstream traffic (uploading to facebook or youtube, etc.).

I recommend a firewall that has traffic shaping capabilities so that you can insure the upstream bottleneck is at your router so that you can prioritize your SIP traffic and ACK packets. I use pfsense. It is free and powerful, but it has a bit of a learning curve, and the documentation sucks (unless you buy support, in which case you get a book).

I've also looked at Sophos UTM which is free for under 50 IP addresses. It looks pretty slick, and may handle some of your other firewall needs like web filtering and virus protection.

And Igaetz is right, faxing using anything other than a dedicated POTS line and a real fax machine is like a bleeding ulcer - the pain just never ends.
 

Robert-BCC

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PIAF without Incredible PBX does work nicely as an inbound fax server as well.
Good to know. I’ll start with the POTS lines for fax and consider migrating someday…

I don't know if Vitelity offers failover service, but if they do it's just a setting in their portal.
They do, and they even have a nice video describing how to set it up.

>What router are you planning to use?
Well at home I have a SOHO running DD-WRT and any firewall setup info would be appreciated (I got the NO PORT FORWARDING message). For the actual install, Cisco or Juniper, we’re working on getting a donation. We're getting static IPs for both ISP connections.

> PIAF purple is great, but if setting up a new production server today you want to be on green, ver. 11 is the current LTS asterisk version.
Reinstalled as Green + v11. It would be cool if Green was the default PIAF style during the install process. Yeah yeah, I should have read the readme that accompanied the ISO...

It sounds like your ISP connection is going to be DSL.
Only the voice traffic will be on the DSL link, but QoS on the secondary link could be useful for backup purposes.
 

atsak

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Juniper is much easier to configure for SIP than Cisco, and an SSG5 will do fine for you, which is $125 used. In short:
Disable the SIP ALG
MIP the static IP and add a policy to port forward the RTP UDP stream (10 000 - 20000) from ANY
Add another policy to allow the SIP UDP (5060) from ONLY Vitelity (and any other provider).

Run this way for years no hack problems. Junipers are strict firewalls and you must port forward for it to get two way audio.
 

rjaiswal

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I have had good luck with ubiquiti edge router lite routers. Have a bunch deployed in front of local asterisk boxes. They work like a charm. And for $99, it's a steal.
 

Robert-BCC

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Juniper is much easier to configure for SIP than Cisco.
We have a volunteer with Juniper certification so we're probably headed that way.
MIP the static IP and add a policy to port forward the RTP UDP stream (10 000 - 20000) from ANY
So RTP is the actual payload, and we're expecting this traffic from the initiating station (or their proxy), not my SIP provider, thus ANY. Is that correct?
 

rossiv

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We have a volunteer with Juniper certification so we're probably headed that way. So RTP is the actual payload, and we're expecting this traffic from the initiating station (or their proxy), not my SIP provider, thus ANY. Is that correct?
Depends on the provider. I think you earlier mentioned Vitelity. I am pretty sure that they do proxy media through their servers so you would need to allow RTP from their IP.
 

jeff.h

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I haven't seen too many providers that don't proxy RTP, they don't want you to know who they are buying from or else you might go directly to the source and they lose their markup.

As far as Juniper being easier to configure than Cisco, I can't speak to that other than I have used a ton of Cisco gear from small business appliances to Nexus 5k and everything thing in-between and have found that anything is easy once you know how it works :D

I recently had to work with someone who is rather tech savvy, but not network savvy on an issue with their local router that was not sending packets out to my PBX on the right ports. It kept wanting to use random high ports instead of 5060 no matter what we told it to do and went through a couple different models that he had on site, swapping them out to see what would happen (I'm looking at you Netgear and Apple Airport Extreme) He ended up getting a Cisco Small Business RV320 and it took him all of about 15 minutes from start to finish to get it setup all on his own with no help from me and it works like a champ with SIP.
 

w1ve

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BTW, Vitelity hosts PBX-in-a-Flash for $19.95 a month for a basic server, which will probably suit your needs. This puts the a hosted server right next to your trunking provider (<1mS ping time). I have some clients using it and so gar it is working very well. The client has Comcoast Business to the net (100/25) and Cisco routers with vlans for voice, and have about 100 extensions at the moment. Vitelity does not care who you trunk via, so you can have backup with anyone.
Something to consider. I'm running a few production PIAF servers for small offices on Digital Ocean $5/month 512K droplets.. .about 4 months now and flawless.
 

Robert-BCC

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BTW, Vitelity hosts PBX-in-a-Flash for $19.95 a month for a basic server, which will probably suit your needs. This puts the a hosted server right next to your trunking provider (<1mS ping time). I have some clients using it and so gar it is working very well.
Yeah, Vitelity has a good video about this, and I love the proximity of trunk to PBX, my only concern is the latency. Breast Cancer Connections is in Palo Alto (near San Francisco) and Vitelity only seems to host in Denver. Are your clients similarly distant? Has latency between their phones and their PBX been an issue?
 

w1ve

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My clients are in Detroit, MI. Running about 35-40mS latency... really not an issue. Perfect call quality.
 

w1ve

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I'm pinging nike.com (Palo Alto. CA) from a server @ Vitelity.
PING 66.54.56.30 (66.54.56.30) 56(84) bytes of data.
64 bytes from 66.54.56.30: icmp_seq=1 ttl=244 time=39.7 ms
64 bytes from 66.54.56.30: icmp_seq=2 ttl=244 time=41.2 ms
64 bytes from 66.54.56.30: icmp_seq=3 ttl=244 time=100 ms
64 bytes from 66.54.56.30: icmp_seq=4 ttl=244 time=125 ms
64 bytes from 66.54.56.30: icmp_seq=5 ttl=244 time=39.3 ms
64 bytes from 66.54.56.30: icmp_seq=6 ttl=244 time=39.5 ms
64 bytes from 66.54.56.30: icmp_seq=7 ttl=244 time=39.6 ms
64 bytes from 66.54.56.30: icmp_seq=8 ttl=244 time=39.9 ms
64 bytes from 66.54.56.30: icmp_seq=9 ttl=244 time=39.6 ms
64 bytes from 66.54.56.30: icmp_seq=10 ttl=244 time=39.6 ms
64 bytes from 66.54.56.30: icmp_seq=11 ttl=244 time=40.5 ms
64 bytes from 66.54.56.30: icmp_seq=12 ttl=244 time=40.1 ms
64 bytes from 66.54.56.30: icmp_seq=13 ttl=244 time=40.1 ms
64 bytes from 66.54.56.30: icmp_seq=14 ttl=244 time=39.6 ms
64 bytes from 66.54.56.30: icmp_seq=15 ttl=244 time=42.0 ms
64 bytes from 66.54.56.30: icmp_seq=16 ttl=244 time=39.9 ms
64 bytes from 66.54.56.30: icmp_seq=17 ttl=244 time=39.5 ms
64 bytes from 66.54.56.30: icmp_seq=18 ttl=244 time=39.5 ms
64 bytes from 66.54.56.30: icmp_seq=19 ttl=244 time=39.5 ms
64 bytes from 66.54.56.30: icmp_seq=20 ttl=244 time=39.6 ms
64 bytes from 66.54.56.30: icmp_seq=21 ttl=244 time=39.6 ms
Other than the odd 100-125, looks good.
 

Robert-BCC

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OK, but I will do better than that Ward. If I can get this thing up and running, I'll contribute the difference in their monthly bill to the PIAF project from my own wallet.

Robert
 

w1ve

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Robert -- you won't go wrong going PIAF. Best support in the biz -- a lot of it right here on this forum. And if you really get in the weeds, the guys who wrote much of it are here as well.

Ward -- I've used the link for my vmobile service. My client went to vitelity before I could give em the link, but I'll get ya on the next one.
 

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