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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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Ya'all are asking the wrong question How much are you willing to spend?
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May 27, 2016 15:15
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Pretty normal usage requests, so nothing jumps out as an obvious choice Mitel: has the conference system built in for setting up and hosting(dial-in, PINs, etc), remote phones are easy, options for hot desking/users with phones at home and office, able to link systems or use one and setup the other side as remote phones. Assuming for your conf rooms you have polycoms or something similar which is no problem(either analog or SIP). Runs on mitel hardware, standard servers, or even as a VM, or as a hosted solution. Systems would be the MiVoice Business, used to be the MCD/3300 Toshiba is similar but items like the conferencing are addons. Runs on dedicated equipment or as a hosted solution. This would be the IPedge(hardware) or VIPedge(hosted) Allworx has a recent system out that goes upto 180 users but that's new since I left the last place I was at, was still working on the 6x and 48x lines
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May 27, 2016 16:05
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Harik posted:Out of curiosity - I run a PBX that does accept remote SIP so people's cellphones can have SIP clients ring at the same time as their desk extension does, plus road warriors. Saw this on one of my mitels that I was adding sip trunks to. Just try after try after try to register a sip extension. All random extensions all getting 403forbidden There were about 5 different ips they came from. China or Russia I forget which internet hellhole they were from.
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Mar 8, 2017 20:45
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Harik posted:Charter/spectrum has started molesting SIP traffic, has anyone else seen that? The packets arrive, but they're rewritten to come from some not-us address, then AT&T Uverse used to(maybe still does?) a thing where their modem/gateway/phone device would hijack traffic on 5060. SIP phones at a house behind one of these gateways would connect but after that any calls and other traffic for the phone would just be poo poo. This was happening even with Uverse houses without phone service, it was some default provisioning in the network, so maybe thats an avenue to check
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Mar 15, 2017 15:20
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