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There was a VOIP thread but it got archived. I have a small (5 employee) office and our phone system is basically a Grandstream PBX from 2008 and 5 Grandstream IP phones. It all works perfectly fine in our office but one of our employees is going to be working offsite for the forseeable future and needs to be able to connect remotely. Other than spending too much time on the computer and above-average Google skills I don't know jack poo poo about computers so I've been having a bitch of a time trying to get remote extensions to connect to the PBX and it's all most likely due to NAT issues. And honestly I don't feel comfortable opening up ports on the router to allow internet access to the PBX because it's inevitably going to get bruteforced and used as an Indonesian call-center or whatever. Unfortunately for me, due to the nature of our work, I can't have the offsite employee making calls from his personal phone because a) It would get expensive, and b) our clients are olds and change makes them nervous. I also need to keep our current phone numbers (see b) above) so changing our SIP provider is not an option. I'm not in the US so assume # portability is not an option. What's the easiest way for me to either a) somehow jury rig my current PBX to to accept remote connections without waking up one morning to a $10,000 bill, or b) use some sort of PBX service on the cloud that allows me to use my current SIP provider instead of hiring theirs. I'm willing to pay a monthly fee if it means not having to set up and manage my own Asterisk server.
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# ? Apr 9, 2016 04:47 |
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# ? May 13, 2024 08:39 |
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Some kind of site to site VPN, preferably limited to just the phone. I don't have any particular suggestions, but that is what you need to look into.
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# ? Apr 11, 2016 00:13 |
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Do one of those webinars that gets you a free Meraki Z1 or MX appliance and use that to build a tunnel back to your office.
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# ? Apr 11, 2016 01:32 |
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Two things offhand I can think of. 1 - if you don't want to mess with site-site VPN shenanigans, just make sure this user has a static IP address on their home connection, or can order a static IP address (you can then whitelist their IP to talk to your pbx). Even if they can't, and are on say, a coax cable connection, you could whitelist their /24 netblock and typically even if the are dynamically assigned another IP in 4 weeks or when they reboot their router, your whitelist will still allow things to work. Disclaimer - I've had to do this with a few one-off users on a PBX solution we manage and it has worked well enough. 2 - You could put them on a separate system, something like RingCentral single standard user. It's like 35 bucks a month for 1 user. You could then just set up a virtual extension or forward in your PBX so internal staff could dial his extension or 10 digit DID and reach his phone. Your remote user wouldn't be able to dial the Office-PBX extensions unless you set up some type of speed dials on the remote side that would dial DIDs back into the system. It's kind of hacky but if this isn't scaling up at all, this setup would work without you having to open up any ports on your firewall. I'd say a softphone is feasible but if this is a technically illiterate remote user it may be too much headache, plus troubleshooting softphones adds the overhead of "oh gently caress my PC microphone sucks on my laptop." and "oh my PC can't make calls help malware and PUPs out the rear end!" 3 - another thing to caution you about with remote users is "ghost calls" whereby a lot of the default configs on phones allow direct IP calls. You'd want to check your Granstream configs for this. What this means is that remote user's phone just rings and rings randomly off the hook at 3 AM because his lovely Linksys firewall is being scanned for SIP (5060). A lot of the NAT implementations on these consumer-grade firewalls are not very restrictive and will NAT through these scanners and send SIP Invites to the user's phone, thus causing it to ring with junk caller-ID at random hours. Google "sip ghost call" for more info. Solution is typically disabling "allow direct IP call" on the phone via its web UI or config files. 4 - just migrate everyone to a cloud-based PBX solution so you don't even have to think about your on-prem PBX anymore
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# ? Apr 11, 2016 05:20 |
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Important question: is this dude just working from home or is he a road warrior?
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# ? Apr 11, 2016 05:47 |
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I use VPN with a softphone on the PC, it gives the same extension as the wired phone at the office. If you can setup VPN, GrandStream should have an app for phone or client for PC that'd work.
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# ? Apr 12, 2016 02:17 |
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I'd really rather use a hardware phone, since this is not a technical user and he will be working almost exclusively from home. I've been mulling it over and I think the "easiest" solution will be to get him a router that supports ddwrt and have the router OpenVPN into the office network. Then I just hardwire the phone to the router and it should connect transparently to the PBX at the office. Not the cleanest solution, but hopefully it'll work. I think I have spare router that I could use to try it out.
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# ? Apr 12, 2016 05:55 |
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An ASA5505 is like $300. Don't fart around with DDWRT in a business environment.
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# ? Apr 13, 2016 11:50 |
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An edge router from ubnt is $50 for the low end.
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# ? Apr 13, 2016 13:42 |
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FatCow posted:An ASA5505 is like $300. Don't fart around with DDWRT in a business environment. gently caress it,
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# ? Apr 13, 2016 23:30 |
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Morganus_Starr posted:gently caress it, What do I do with that Link-Skis thing you mailed me?
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# ? Apr 15, 2016 04:25 |
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gently caress me I've been trying to get ddwrt to connect to my Asus router via OpenVPN for the better part of a day now and it's no use. Probably gonna end up buying another Asus router just for this.
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# ? Apr 15, 2016 04:40 |
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Ur Getting Fatter posted:gently caress me I've been trying to get ddwrt to connect to my Asus router via OpenVPN for the better part of a day now and it's no use. FatCow posted:An ASA5505 is like $300. Don't fart around with DDWRT in a business environment.
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# ? Apr 15, 2016 14:43 |
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# ? Apr 15, 2016 14:57 |
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Methylethylaldehyde posted:What do I do with that Link-Skis thing you mailed me? so dat linskees box is my telephone????
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# ? Apr 15, 2016 17:05 |
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Super Slash posted:so dat linskees box is my telephone???? what? Oh the phone has to be plugged in to my router? I just plugged it into my laptop - these things should just work! UGHHH
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# ? Apr 15, 2016 23:30 |
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I appreciate the suggestions but I'm not in the US, getting one of these will be more of a hassle than it's worth, not to mention I'll have to pay almost double what it costs for you guys. This isn't mission critical plus the employee will only be doing this for a few months. I'm gonna see if I can solve this on the cheap, worst case scenario he has to use his home phone it's not a big deal.
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# ? Apr 16, 2016 01:06 |
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Super Slash posted:so dat linskees box is my telephone???? You just triggered my "used to work third tier support at Vonage PTSD"
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# ? Apr 16, 2016 02:09 |
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I know you said you don't want to mess with your firewall, but it is literally 2 ports that need to be forwarded; defaults are 5060(UDP) and 10000-11000(UDP), both changeable in the PBX settings. Like was mentioned above, for the truly paranoid, you could whitelist his home IP and apply that to the firewall/NAT rules. Or find someone that does hosted SIP, port your numbers there, and just re-use the grandstream phones How do I setup a remote extension to the UCM6100 series IPPBX? (yeah this might be a different PBX model than you have, but the logic is still the same) If the UCM is using an external IP (not behind a NAT), then you don’t need to configure anything for remote extensions, but if it is behind a NAT then these are the steps: 1. Navigate to PBX-->SIP Settings-->NAT on the UCM6100's web UI. Put the external IP of the network in the field “External IP Address” (if a domain is being used instead. e.g. DDNS, then use the next field “External Host”) and the internal IP of the UCM in the field “Local Network Address” 2. Port forward in your router the SIP port for the UCM (by default UDP:5060 and can be changed under the “PBX” > “SIP Settings” > “General” tab –we recommend changing it to increase security) and the audio ports (by default the range UDP:10000-20000 and can be changed/decrease under the “PBX” > “Internal Options” > “RTP Settings” tab – we recommend decreasing it to 10000-11000). 3. On the remote phone(s) use the external/public IP of the UCM as the SIP server. Also put in the SIP User ID, Authenticate ID, SIP Password for the remote extension. 4. On the remote phones you may want to enable “Keep-Alive” for NAT settings. In Grandstream phones the option is “Auto” for the setting “NAT Traversal” located under the “Accounts” > “Accounts #” > “Network Settings” tab of the phone’s Web interface. If those options do not work then select “STUN” and put “stun.ipvideotalk.com” in the field “STUN Server” located under the “Settings” > “General Settings” tab also of the phone’s Web interface. 5. On the phone(s) you may want to enable “Use Random Port” by setting it to “Yes” under the “Settings” > “General Settings” tab of the phone’s Web interface. For this setting you need to reboot the phone to take effect.
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# ? Apr 28, 2016 20:00 |
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Ok legit question; Currently we use Salesforce along with a hosted VOIP provider and both are connected. This year we'll be moving away from both as a bespoke CRM build is currently underway and we'll be moving to a much cheaper VOIP provider. Now the problem; our director wants to move to the new VOIP while we're still using Salesforce. Why is it a problem? Because the provider doesn't have any kind of connector, adapter, or call center app for Salesforce. So I'm shopping around the SF app store without much hope as it's filled with expensive businesses to sign on with, or cruddy outdated packages which are highly questionable. Any recommendations out there for a connector?
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# ? May 5, 2016 16:26 |
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From a brief look and not a recommendation because I have never used it before: https://appexchange.salesforce.com/listingDetail?listingId=a0N30000004g5sgEAA If your new hosted provider doesn't have a TAPI/CTI thing to connect to then you're probably out of luck.
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# ? May 5, 2016 16:31 |
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Ive been selling and installing SpliceCom phone systems for years. Reliable and cheap!
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# ? May 7, 2016 00:21 |
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I've just had an email that one of our tenants has been accepted for the Skype for Business PSTN Calling trial (we're in the UK so it only just got here). Is anyone else using / testing this out as a replacement for a phone system? Any tips on what to avoid?
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# ? May 10, 2016 08:35 |
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Thanks Ants posted:I've just had an email that one of our tenants has been accepted for the Skype for Business PSTN Calling trial (we're in the UK so it only just got here). I beta tested it on a fairly small scale and had no issues with it. How big is your organization?
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# ? May 10, 2016 17:15 |
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This particular one is only 60 seats.
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# ? May 10, 2016 17:26 |
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Even if you ignore everyone having a mobile a decent voip system is nice for screen sharing and what not. Of course there are plenty of lovely ones out there, looking at you Cisco. Best one I have used is Skype for business which is shocking that MS didn't gently caress it up.
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# ? May 13, 2016 16:59 |
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Thanks Ants posted:From a brief look and not a recommendation because I have never used it before: I actually ended up giving this a go since there's pretty much nothing else, unfortunately it was waaaaaay too fiddly to use. You need to install and configure a TAPI driver on every machine, along with a local client of the connector software, the worst part is Salesforce has it's own built in crappy call center you have to use which is missing important parts (e.g. transfer call list). So ultimately things would get even more complicated and I'm sure our users would hate me off the face of the earth. No beef with you but just a general gripe. Personally I don't think there's any point whatsoever in jumping providers for the sake of a few months in the first place, but I've told the boss it's just not practical and we'd be worse off for many different reasons.
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# ? May 13, 2016 17:07 |
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BigPaddy posted:Even if you ignore everyone having a mobile a decent voip system is nice for screen sharing and what not. Of course there are plenty of lovely ones out there, looking at you Cisco. Best one I have used is Skype for business which is shocking that MS didn't gently caress it up. They've hosed up the Windows client because it cannot cope with multiple displays running at different DPIs. You can take a Surface Pro, connect an external display, drag the Skype client onto the Surface display and it becomes impossible to see what you're doing. This is a product they are really pushing, running on their latest OS, running on their flagship hardware.
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# ? May 13, 2016 18:52 |
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BigPaddy posted:Even if you ignore everyone having a mobile a decent voip system is nice for screen sharing and what not. Of course there are plenty of lovely ones out there, looking at you Cisco. Best one I have used is Skype for business which is shocking that MS didn't gently caress it up. My company's Skype for business deployment is a clusterfuck and we all hate it.
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# ? May 13, 2016 23:35 |
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Looks like Meraki is getting into the VOIP game. https://meraki.cisco.com/products/communications
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# ? May 17, 2016 17:47 |
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I would be massively interested in a free one from a webinar. gently caress phone systems.
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# ? May 17, 2016 17:56 |
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Moey posted:Looks like Meraki is getting into the VOIP game. Their initial provider is offering (what seems to me) to be fairly inexpensive service compared to the competition. I'm not sure if they've publicly revealed pricing yet (we're a partner), but the handset is certainly a premium item right now, hopefully they offer cheaper options down the road.
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# ? May 18, 2016 01:18 |
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It makes sense for them to launch with the most flashy option they can, be good to see phones with buttons though since I hate touchscreens on desk phones. I'm surprised at how much duplication of effort Cisco and Meraki are doing, presumably there is a desire to keep the two brands separate but it seems really wasteful. Cisco have Spark, handsets etc.
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# ? May 18, 2016 01:29 |
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Thanks Ants posted:I'm surprised at how much duplication of effort Cisco and Meraki are doing, presumably there is a desire to keep the two brands separate but it seems really wasteful. Cisco have Spark, handsets etc.
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# ? May 18, 2016 03:42 |
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Thanks Ants posted:It makes sense for them to launch with the most flashy option they can, be good to see phones with buttons though since I hate touchscreens on desk phones. There is no real coherent strategy or narrative from cisco with collaboration from what I can gather.
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# ? May 18, 2016 06:23 |
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Preface: I've never used Asterisk/VOIP stuff ever and I have no experience with it. Wanting to set up something that has a bunch of different VOIP lines tied to it that will all dial an 800 number at a specified time, and then when one them gets answered it will connect me then drop all of the other calls. Any links/recommendations/tutorials for a good starting point?
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# ? May 18, 2016 20:24 |
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I installed AsteriskNOW on a VM to play with and goddamn, I'm sorry for everyone that has to cj this professionally, you are not paid nearly enough.
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# ? May 25, 2016 14:43 |
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Ur Getting Fatter posted:I installed AsteriskNOW on a VM to play with and goddamn, I'm sorry for everyone that has to cj this professionally, you are not paid nearly enough.
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# ? May 26, 2016 04:43 |
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adorai posted:The learning curve is very steep, but once you get the hang of the product it's not as terrible as it seems at first. In the end most of my problems came down to NAT issues as usual, but yeah, it seems to be mostly working now. I just wish it had better SIP error logging (why the gently caress do I have to manually enable SIP logging through the CLI every time I reboot?). Also I had to literally reinstall the whole server three times because trying to upgrade to the newest version kept loving dependencies in new and exciting ways. Other than that, it's chock full of really good features AND it's free so actually I'm sorry for my bitching Asterisk devs
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# ? May 26, 2016 15:28 |
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# ? May 13, 2024 08:39 |
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I just wanted to say that as a network guy who's been forced to become our VOIP guy, the thread title caused a small tear to show up in the corner of my eye.
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# ? May 26, 2016 17:01 |