I'm searching for some insight into the SV8100 and the DT700. Currently for remote use cases I have access to the SIP server established through a IPSec VPN tunnel. I've been tasked with finding out if it is possible to have the DT700 connect directly to the SV8100 without the usage of a firewall at the remote end. I personally prefer requiring the usage of a VPN tunnel but from a usability standpoint for users it is easier to just plug the phone into a network jack and let it connect. I would be okay with allowing a specified, static public IP address through my firewall to access the SV8100.
so what I'm really trying to figure out is what I need to change in my SV8100 and a DT700 in order to get the DT700 to connect.
Would I need to activate DHCP on the SV8100? Would doing so have an impact on local IP phones that have DHCP handled by my firewall? Can I just specify in the DT700 a static IP address to use?
What's the best approach to getting the DT700 connected remotely without the usage of a firewall?
As a disclosure, I dislike the SV8100 but it serves its business purpose so this is what is in use. I personally run 3CX in the cloud for my home VoIP setup and don't have any issues with it (wish it was an option to switch out the SV8100 with).
I'm looking for a self or cloud hosted VOIP solution that will allow my 100% mobile workforce to be able to make outgoing calls from their cell phone using cellular voice (not just data). My workforce sometimes encounters areas where cellular data isn't great, but cellular voice works fine. We want the caller ID to still show up as our main office number, so just making an outgoing call from the cell phone using personal caller ID won't work.
Wish me luck!
Moving from 3CX to FusionPBX. Having not touched anything other than 3CX I guess it'll be an experience, However, It Can't be that much different in terms of setup and configuration, I think the biggest hurdle will be getting Fusion to work with my mobile devices, finding a good app. A good desktop App that would support the Equivalent CTI mode in 3CX Phone would be good also to work with desk phones.
Any advice - greatly appreciated!!
So I posted this over at the Cisco subreddit but didn't get very far so here I am.
We currently use a VOIP system that is mostly digital Polycom based devices with a few interspersed analogue devices connected to Audio Code boxes. I migrated our on-prem Mitel PBX to a hosted solution with 8x8 not to long ago but I still have 3 devices on the old Mitel, namely our barrier/gate controls. So the way they work is we have an intercom at our staff entrance and if a non-staff approaches and presses the buttons on the intercom they are connected to one of 3 users. The user confirms who they are and then presses 3 on their phone to let them in via the gate.
So I have these Cisco SPA 122/112 devices that I know can be configured with a dial plan. I have setup a test one and I can get it to auto-dial up in my office but when I connect it to the gate patch port it appears as if the line is always open in that it will just constantly dial my extension (using this as a test so as not to bother people). I have tried the ( Px <:xxxx>) configuration and also the ( S0 <:xxxx>) configuration but both do the exact same thing just one with a pause and one without.
Has anyone implemented the Cisco SPA 112/122 into a usable intercom/gate entry system? I have rummaged around the Cisco forums and people have said they have done it but not explained how, along the same lines as this link:
Any help would be much appreciated. I don't want anyone to do the work for me just a nudge in the right direction if you know it! Thanks.
Anybody using voip on IPV6. I have a self hosted Freeswitch server which is working well with IPv4. My ISP does provide Ipv6 and thus wanted to use ipv6 profile. The major advantage is that it will do away with NAT. Gs Wave supports Ipv6 but prefers it over IPv4. Anybody know how to force it to use ipv6?
Whenever i call from Linphone SRTP is used, however whenever i receive a call SRTP is not used. Can anybody confirm they can receive SRTP calls?
I took a sip trace of the call, and it seems that Linphone never offers the SAVP flag for SRTP. I do have SRTP set under Media Encryption in Calls.
I am testing the calls using my own Freeswitch server. The same configuration works fine Zoiper, however it doesnt support ipv6 which i like to get going.
I've got an Obi200 VoIP box and my outgoing voice data is compromised (I sound jittery or they can't hear me at all) if too much other uploading is going on (ie, cloud backup, etc). There was fairly easy-to-use QoS settings on my old Buffalo router running Tomato, but the OpenWRT (LEDE) currently on my Linksys does't seem to have anything similar. (I did install the SQM QoS script and set it to cake/piece_of_cake, but it doesn't seem to be helping at all.)
Can anyone recommend a firmware that would provide some reasonable QoS settings? DD-WRT? Original Firmware? Is there something newer I'm not aware of yet?
I'm looking to buy two CP-7940g's on eBay. I want them to work like this. I'll have two phone numbers and when someone calls both of the phones will ring.
Also, can I name lines? "ABC" and "XYZ" so I know which number is being called.
Which distro/program should I use?
Thank you for the help
Hi, everyone. I was browsing web, but couldn't find instructions on how to setup open source SIP server that allows new users to register their own account (from web or sip clients). If someone can point me in right direction or give any tips.
We have Skype for Business and we were using Polycom VVX3000 conference phones. Since they're old and going to be unsupported soon, we're moving to the Trio 8500 phones. Since deploying a few of these, we've had users complain about them picking up tons of ambient noise and callers being choppy because of it. Like, you can't hear someone on the phone unless you mute the phone in the room.
I know it's a newer, higher quality mic, but good lord. We've tried with and without the noise cancelling setting on the phone, but it makes no real difference. We do have a config file on an FTP server, but it basically just enables the Skype for Business backend instead of the Polycom one.
Any ideas? I reached out to our vendor for help but they just want to add more mics, which seems ridiculous in a six person conference room. Oh and Polycom basically said "we don't support this" when I put a ticket in. So that was fun.
I would like to use (https://www.ebay.com/itm/Linksys-SPA3102-Phone-Adapter-with-Router-1-FXO-1-FXS-VoIP-phone-adapter-Gateway/272668192787?_trkparms=aid%3D555019%26algo%3DPL.BANDIT%26ao%3D1%26asc%3D20150817211709%26meid%3De5a25a8163424a5b94630e998158279c%26pid%3D100506%26rk%3D1%26rkt%3D1%26%26itm%3D272668192787&_trksid=p2045573.c100506.m3226)[this device] to route my ipphone's calls over my regular telephone line.
Will this work?
And if so, is there anything else I'm going to need other than my telphone line and a router.
Also my line 1 and 2 come over the same telephone line. Is that going to be a problem?
We're implementing a new phone system this summer, and I have options narrowed down to Mitel MiVoice, Cisco Business Edition 6000M, and Avaya IP Office. We're looking at about 500 handsets, we have private fiber connecting all of our buildings...overall pretty straightforward implementation. Looking for opinions and experiences with these systems, the administration interfaces, support experiences (which I know in some cases has more to do with the reseller than the manufacturer), etc.
Who likes what? Why don't you like the others?
(Edit: Coming from an Avaya Definity system dating from 1998. T1/PRI for connectivity, no call center functionality.)
I have a FreePBX instance hosted at our datacentre on a 1:1 NAT. My customer has a Ubiquiti Edgerouter Lite sat on a 50mbit/s leased line at his premises. They have a ring group that's set up to call 16 handsets on site when a specific inbound route is requested and will ring for 12 seconds before handing off to a third party answering service.
I'm seeing reports of sporadic calls going to the answering service when none of the handsets have rung in the office, yet the FreePBX server indicates it tried to ring each of them and it has the call recording from when it went to the external answering service.
Could anyone shed any light on why a whole 16 handsets would fail to be rung at one time? or what I could look at to try and fix this please?
I just got Magic Jack. People say they can hear me fine. One person said i sounded a little bit far away but was clear. Another person said I founded totally clear.
But on my phone ( I got a refurbished Vtech CS6124) I hear slight sizzle sound a lot of the time. I use Cox Cable and have the 20 Megabit plan. Do I need a DSL filter hooked into the phone? I have the Magic Jack plugged into my Asus router. Are there other things I can try?
Hello! I've been trying to test out a voip system here in my office. My buddy had an extra OBIHAI 1062 IP Phone. So I figured I'd give it a try to test here. I know it's "unsupported" according to their website. I'm running 3CX locally on my machine. I just don't know how to get that phone to connect to 3CX. I already have 3CX registered with Callcentric. But having a hard time with 3CX and the Obihai Phone. Any help would be appreciated!
We have an asterisk server and we are running into a problem with our click to dial from our computer. How it works is the user clicks the number, their desk phone rings and when they pick up, it dials the number they are trying to call. Our issue is that these show up as incoming calls on the phone and in our CRM system vs. outgoing. Does anyone have experience with this and found a solution?
Is there anyway to change the MAC address on a Yealink 46g? I'm deploying 25 of them and need to change the mac to make them work with my provisioning server.
I have a virtual phone number which is forwarded to my mobile. This mobile is almost constantly on. However, on the telco's providers online portal - there seem be a lot of calls being classified as "unanswered" and then getting directed to "voicemail". These calls are not registering as "missed called" on my smartphone. I have tried testing this setup using my own landline and it always works but I am not sure how well it works from other networks? Could there be an issue here?
We're trying to set up a Call Center, we'll be using FreePBX and NIPO CATI surveying system, and wanted to see if anyone can recommend a free auto-dialler?
Any insights on CATI, FreePBX or free auto-diallers, will be welcome too.
What is the difference between the Polycom VVX500 and VVX501?
I've waded through the generic sales blah-blah on their web site and the VVX500-series PDFs that lump them together. No clarity emerged. I guess they couldn't be bothered to provide a feature matrix chart.
Thank you in advance.
I bought a set of Yealing Sip-T20 phones. It turns out Sip-T20 is not a POE phone even though it says "POE optional" on the box. Unfortunate I cannot return them since this happened on craigslist.
Since I do not have the space for another wall plug I wanted to power the phones with POE from my POE switch. I looked at the phone and the phones takes 1.2amps at 5V. Am I able to plug in an adapter like this
I am trying to figure out how many volts does each port supplies on the switch
So this is for the old school folks out there. Need an alternative to the 4303 dacs that we have in production because lets be honest, we hardly use them and the interface is terrible.
We have a migration where the DS1 terminations to an STP in our datacenter have to be relocated to a new facility and we were hoping we could use this opportunity to upgrade some of the equipment. Anyone know of alternatives to the 4303 that we could potentially use?