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Posted by
FreePBX Developer
1 month ago
Stickied post

Unfortunately, when you go to it's redirecting you to without any hints as to where to go, rather than (In my opinion) the correct place of

I expect we'll have a bunch of people going 'The FreeSwitch wiki is broken!' so, to head that off, please go to instead and then search inside confluence for what you want.

Sorry 8-(


My company is migrating from an on-prem voip system to one hosted in the Internet. (Don't really want to say any names, but it's one of the big ones that gets mentioned here a lot.)

Our number one challenge has been widespread call quality complaints. It's reached the point where our users at the branches use their personal cell phones exclusively to conduct business, while the shiny new desk sets just collect dust.

The problems mentioned are specifically "audio cutting in and out, distorted audio (under water sound), or garbled audio (robot voice)."

Our connectivity is that we have SD-WAN boxes at each branch, with broadband connectivity handed off to them... usually just typical business grade DSL or Cable Internet.

Now, before you go too crazy about blaming the SD-WAN, know that the existing on-prem system is good as gold. There are no call quality complaints on that system. Everything works beautifuly.

We've tried letting the SD-WAN box work its magic with the forward error correction it can do, and it doesn't solve the problem. So we've also tried split-tunneling the new phones directly out the broadband at the branch, to give it the shortest path to our hosted provider, they still complain about the same issues.

At this point we're really starting to wonder: is it just them? I know their stuff works a little different than traditional voip. They anchor media for all calls, so if you have two desk sets in the same branch, in the same vlan, their Media Stream still goes all the way out to the cloud and comes all the way back in. Traditional voip systems that wouldn't happen, call setup would be to the "cloud" but the RTP would be directly phone-to-phone. I don't know why they don't do it that way.

We're kind of kicking around the idea that this is the reason the quality sucks because every rtp leg has to go out across the Internet and come back, that it's just not going to be perfect, but of course no one wants to accept that answer.

So I'm posting here to you experts hoping we can find a more scientific approach to figuring out specifically where things are breaking and what we can do to fix it. We have a lot of tools at our disposal, we have the ability to take remote packet captures at the branch and at the data centers and Internet egress points. We have end to end qos set up and verified everything is marked everywhere, except for our main Internet pipe which our ISP is stripping tags for some reason. (Still fighting that out with them.)

I've found taking pcaps of calls to be less helpful than I thought it would. It varies from "ok this is showing minimal jitter, no packet loss, and a MOS score of 4+... yet they complained about horrible quality" to "yep I can see problems" but not giving any insight into why and how to fix it. Ugh.


Is there a general consensus of what the top-quality BYOD voip providers are at the moment?

I'm discoing my POTS svc and am switching over to VOIP. I've heard good things about Callcentric and some others -- what're the best-in-class providers at the moment?


Apologies if this is the wrong sub, but does such a device exist... an office speakerphone I can receive and place VOIP calls, that will also allow me to pair my iPhone to use as a speakerphone with also.


Hi all,

I'm hoping that I've come to the right place. I've recently come across two now discontinued pieces of equipment that I'd like to play around with and explore what features are available. Unfortunately I don't have any documentation with them and I'm hoping somebody here could help me out. I'm looking for any and all documentation available for

  • StarLeaf 6010 Teleconference PBX
  • StarLeaf 6310 Teleconference Controller

Thanks in advance!


Pretty much as the title says. Long version:

Bought an Obihai 200 last month and bought incoming and outgoing service with Callcentric, to be a "landline" service after relocating. All seemed well with testing over a week or so, and we gave the number to everyone and I put another $20 in credit in the account.

Saturday last, I was out, and tried to call the number for my wife to pick me up from the 'bus station. Out of service. I ended up ringing my sister in law who picked me up almost an hour later.

I contacted Callcentric who first replied that they were "currently investigating the issue with our provisioning department; you will be updated once further information becomes available."

Didn't hear anything for 2 days until the next message was that the problem was now fixed. The only thing that approached an apology was:

"We thank you for your patience as we resolved the issue."

I signed up my father with Localphone also last month, and their 25% free credit offer code didn't work. After contacting them, they added the free credit manually and gave another code worth 10 pounds. (I have now signed up with them too, using that discount code).

Callcentric: not even credit for the 2 days (or more) than the incoming number didn't work.

I might think I'm being unreasonable to expect more from a budget provider, but Localphone is also one, and they gave free credit when I didn't even ask or expect it...


We our switching our ISP provided phones we have to a SIP trunking service instead at the end of the month, and am having a hard time getting a solid answer of what it is going to entail getting it setup to work with 3CX. We have PRI lines over fiber with a Rj45 handoff at the moment, and I think we are going to have the same handoff just with a SIP trunk instead. What all is needed to hook the two together? Can I just plug into a ethernet port on the server and be good? Do I need a gateway of some sort? Will it work through VMWare to virtual NIC attached to the 3CX box?

Any help is appreciated.


I'm working with a client that requires VM to text transcription. They are currently setup using SIP service and their voicemail. The voicemail provided by sends out emails with .WAV files attached, but does not offer transcription services. I'm looking for a service that could be used to just automatically send VMs from to be transcribed and then sent directly to the user. I found PhoneTag, but their prices seem a bit high for the low volume of VMs that would be transcribed each month. Any Experience with PhoneTag? Any other suggestions, perhaps even something opensource that could be setup in house?


Hi All,

We are leaning towards moving from Mitel to EvolveIP. So far they seem to talk a big game. Anyone use them? How is their service/support? Any positives/negatives?



Basically, everything goes through when customer 1 makes a call to customer 2 on their other site but customer 1 cannot hear a ringtone even though customer 2 phone rings on their end. I'm hoping someone can cite the reason why this may be and a possible solution?


Does anyone know if you can order the names on the sidecar?

I want to arrange transfer extensions on page page an then a have list of users cell phones on another page.


Hey all,

We have a client who's looking at moving away from their monolithic PABX and across to Skype for Business (Voice) and we're looking at the Sonus SBC1000 appliance to act as their Voice Gateway appliance (connecting out via some SIP trunks from a SIP provider)

Given this is my first foray into Skype for Business (specifically replacing a PABX) I'm hoping someone may be able to answer a few questions quicker than the Ribbon Comms support guys seem to be able to (its taken us over a week just to get hold of someone to talk to in Sales...)


Client had a realllllly old 66block and at some point moved over to this. What type of block is this? Its attached to a really old Lucent Communications Server PBX.


Hi all,

Does anyone know how to use tshark using the command line to only capture SIP and RTP?

Thank you.


looking for ways to spot and debug problems and just analyze in general other then just using wireshark.


After a power outage, all voicemail programming on our Vertical SBX IP 320 was wiped (no passwords, user greetings, etc). Now, when attempting to transfer a call directly to a user's voicemail, the call immediately calls back whoever attempted to transfer it instead of going to the users VM greeting


I'm looking to solve an operational problem I'm facing right now.

Currently I'm hosting Asterisk on site, and I don't really have the knowledge necessary to maintain it. When there's a problem, I can google my way through it but I'm not as fast or as clever as a real VoIP engineer could be. I'm running a shop that used to have a larger sysadmin team than it currently does. Currently a sys admin for the office network, servers, fleet management and do support for 100 - 150 users. I don't think its smart to keep VoIP in the mix.

My fear if I invest time in learning VoIP is that if I spend all this time setting it up, I need to maintain knowledge to be able to respond to emergencies & what not. Any thoughts / recommendations?


Significant outages reported in their Business Voice service this week (big enough to make the news). Is there an RCA more detailed than what the six o'clock news would tell my nana? What happened exactly?


I work in a mid sized business. For practically no reason at all, my boss is convinced that QOS is not applied across the board and he’s annoyed our Network team to the point that they won’t assist.

I want to check it myself. The majority of users are using a soft client rather than desk phone and I want to check if the traffic from the soft client is prioritised.

Has anyone got any thoughts on how to effectively check that our traffic is being prioritised?



I'm looking to send XML request to the Switchvox API.

Digium documentation maybe telling expert how to do it, but an average Joe can't find the way!

So please to point to Digium API page, I have read them all.

My question is, is there any perequise that I need to add to my request ?

https://pbx_ip/xml is working on my network, it ask for authentification, fill it up and I get access and the webpage load what Digium documentation says... the request is empty, this is normal for this

in cURL I use --user myusername:mypsw --data (and here I put the XML request)

Beside that do I need to add more command? like a header/footer command ? a GO and execute the command.

I get error 10000 request empty, or 10002

<error code="10002" message="There was an error parsing your XML request. XML Error : File does not exist: <request at /usr/local/switchvox/lib/perl/Switchvox/ line 2." />

And this is where I'm stuck.

1-Command line is good or not ?

2-What file ? I didn't send a file request with the cURL command ?

3-How do I go browse the content of the PBX harddrive

4 Do I need to turn on something?



Our company recently switched to Nextiva from a ShoreTel 14.2 system, and everything has gone pretty well, except for DNIS passthrough in the call center.

On ShoreTel, we would set our DNIS names and phone numbers on the proper trunk group, and that information would pass through to the destination workgroup. If the destination workgroup was set to roll the call over to another workgroup, the next workgroup would also see that DNIS information.

On BroadWorks/Nextiva, we can set DNIS on one call center, and everyone in that call center will see the DNIS info, but if that call center rolls the call over to another call center, the next call center won't see the DNIS information, only the caller ID / phone number.

This means that if the initial call center is unavailable (someone doesn't pick up, all agents are on the phone, etc.), the roll over call center can't answer the call appropriately. We need this functionality because we have dozens of brands, and all their public numbers reach one call center. Our agents are supposed to answer the phone as if they work for that store, but right now only the agents in the call center with DNIS entries have that capability. The roll over call center has to answer generically, and ask the caller which brand they're calling about.

Is there any way to get around this, or did I set this up the wrong way?


About two months ago I made a post asking for help with migration from copper to VOIP. I was not expecting to get such a an overwhelming positive  response from you guys. Now that I finished the migration from copper to VOIP I wanted to say thanks to the community and to the person who helped me /u/headsupvoip from

I was contacted by a few users here advertising their services. I spoke on the phone to each of them but /u/headsupvoip from Heads-Up IT stood out to me the most. I at first was hesitant to pick anyone because I never knew there are small VOIP providers. To me VOIP is something like a cellphone carrier, and this made me question everyone here. But looking back I am glad I went with Heads-Up IT

What set him apart from others, he was not interested in sellinge his service. He was interested in educating me about the capability of the system. When I was doing my research I was overwhelmed with all these terms, call aperance, call park, virtual extension, IVR, SIP, ALG and so on. No where on ring central was I able to find that information or from other people I spoke to. He also didn't wait for me to ask him about them. He knew me being new I probably never heard of these terms and simply explained them to me. He was passion about making sure I am aware of these things and he wanted to make sure that the system we ultimately setup would be custom for my needs. 

It was only after a week of talking with /u/headsupvoip that we had a conversation regarding signing up for his service. I never felt that he was trying to sell to me. Infact it was me who said "hey can we please have some kind of contract going on here because I want you to know I am serious and not just talking to you."

There is a five star restaurant and then there is a Michelin star restaurant. Well /u/headsupvoip at xxx is Michelin start company. The service, and customer service they provide is simply amazing. Actions speak louder then words. I want you to find me some one at any other company that will log into your router and trouble shoot it until 2am to make sure your system works. This is exactly what happened. We finally finished building the IVR and were ready to go live. I installed all the phone at the end of the day. Put away the old phones and was testing the system, only to discover that some phones were not registering. My hope of having the new system be up and running for the next day were crushed, since this was the end of the working day I didn't expect /u/headsupvoip to solve it. To my surprise user was troubleshooting the system with me. At pm I left the office and he said "hey leave one computer on and I will remote connect to it". He was up and testing the system all the way up to 2am to make sure it was up and running for the next day. I have emails and texts to prove this.

By the way the problem was with my router cisco rv042g (dont remeber the model 100%), which is not a VOIP friendly router. We knew ahead of time that it could possibly cause issues, I even had someone else tell me before i would need to buy a new router for VOIP, well not with /u/headsupvoip

Bottom line is simply amazing I am glad I went with them. There are two types of bills for me. One that I hate paying because they make me feel terrible and then there is a bill that I want to pay. Well Heads-Up IT is the bill I love seeing and paying.


How do I update my Polycom phone firmware in batch? I have >30 phones, a mix of VVX400 and VVX401. Might need a server, but willing to do it to save time.


Does anyone use fanvil g200s ATA devices to hold up old fax machines? We have used those particular devices in the past to run analog phones but they do not work well for fax as they just stop working from time to time.

What do the people of reddit do when it comes to faxing over voip.


I own a two-man business. We opened in 2015 and tried all the providers mentioned above. They all had issues with latency despite us changing routers and getting the QOS settings figured out. We now use Spectrum for our desk phones and auto-attendant and it works fine, but it’s costing us over $150/month. I’m not sure how the system is configured, but it appears to have dedicated wiring to a separate modem .....I think! Not voip anyway. That’s not bad, but mobile apps, three way calling and VM to email would be helpful and we could save up to $100 per month. However, none of that is worthwhile if we still end up talking over each other on our calls due to the delays, or lag in the transfer of data over a voip call. With access to faster internet speeds with Spectrum fiber should we look to try VoIP again? Any advice would be appreciated before I end up back down the rabbit hole!


I'm going to go with several coworkers to demo a potential VoIP solution for our company. I'm not very familiar with VoIP, but I'm trying to learn and understand it better. What are some things I could ask or pay attention to?

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