@RamblingBiped said in SIP Phone registers no audio...:

Here is the guide I followed to get the Yealink T23G to work with OpenVPN: http://www.sunstatetechnology.com/docs/YealinkOpenVPNGuide.pdf

When I disable NAT on the extension the phone is unable to register.

Something is wrong there, that sounds like the VPN is not working as there is no NAT when there is a VPN.

I think that we need a post highlighting the new phones, too!

@JaredBusch said in The High Cost of Unlimited Calling:

@scottalanmiller said in The High Cost of Unlimited Calling:

@NetworkNerd said in The High Cost of Unlimited Calling:

There's also the idea that you can have unlimited concurrent calls and just pay per minute of use while you pay a certain fee for each DID you keep per month. As you said, there is a cap to this "unlimited" number of concurrent calls that gets applied (at least from what I have seen from certain providers).

Sometimes all you can eat is all someone says you can eat.

That's surprising that someone would cap that since normally those are charge per minute and the goal of unlimited concurrency is to encourage you to use as much as possible.

I'm not sure what he is taking about. All the pay per minute SIP trunk vendors I have used have no hard cap on this.

Some like VoIP.ms do put in a soft cap but will raise it to whatever you want.

Their reasoning is to prevent fraud

That I would understand, you don't want someone able to charge tens of thousands of minutes of usage per minute if you didn't know what you were doing. Lots of risk there.

@Veet said in Vendor Harassment Black List:

@scottalanmiller - while I back your intent, I don't think a community based black-list would work ...

Perhaps not, but what if we had a thread and each person maintained their own list on it? Then people could pick and choose quickly from that list?

@JaredBusch said in Voice to Text transcription:

I use https://www.nexiwave.com/

Awesome. I found that earlier this morning and was going to give it a try. What plan do you use for your clients? How is the quality of the transcription? Fairly accurate?

@NetworkNerd & @JaredBusch - We would love to have you both. We can offer you 20% off this year's AstriCon registration with promo code "VUC" at checkout. Otherwise, make sure to check out the Early Bird pricing for next year's event (which ends in late June)

@coliver said in Troubleshooting SIP connection to Remote PBX:

@JaredBusch said in Troubleshooting SIP connection to Remote PBX:

@Romo said in Troubleshooting SIP connection to Remote PBX:

@scottalanmiller Doesn't it do that because iOS doesnt allow any UDP connection in the background? Wouldn't TCP work better in mobiles in order to save battery as well?

ZoIPer has an OMFG kill my battery setting to force it to keep registration in the background.

Which is a good idea to turn off... That thing will eat through your battery in a few hours.

It is off by default, but for people who want a softphone on their iPhone to always be on and registered it is an great option.

@wirestyle22 said in FreePBX Remote Extension Problems:

@scottalanmiller said in FreePBX Remote Extension Problems:

@wirestyle22 said in FreePBX Remote Extension Problems:

@RamblingBiped Hope this is the solution. I unfortunately have Cisco CM and its my first time managing VoIP

So sorry.

Complication for the sake of complication

And spending money for the sake of throwing away profits 😉

@poorizad said in Storing Elastix 2 Call Recordings on NFS NAS Share:

@scottalanmiller hi again....i guess the nfs drive is unmount from the server, because no file record on Nas after the 2016/07/28.....nothing changed on settings.....but a few server shutdown for hardware maintenance.....how can i fix it again ?

Try the mount command again. It might mount right up. If the NFS server has restarted, it would disconnect.

@poorizad said in Storing Elastix 2 VoiceMails on NFS NAS Share:

@scottalanmiller it means i use your previous command with this path . right ?

Correct

@Jimmy_K said in No Outbound calls even TRUNK is registered:

@JaredBusch . Can't be.

It most certainly can be, as that is the error in your log.

@Jimmy_K said in No Outbound calls even TRUNK is registered:

When I use X-lite it is still working well. However when it comes to PBX server, it can't work.

The softphone should not be affecting the dialed number in any way. So whatever digits you dialed there need to be what your trunk is sending.

@Jimmy_K said in No Outbound calls even TRUNK is registered:

They have provide me the prefix 20 + ( international call ) 00 + local number. But it still doesn't work at all

This is not what is in your log. So you screwed something up.

As I said way back in post 5:

@JaredBusch said in No Outbound calls even TRUNK is registered:

This is the call as presented to the trunk.

-- Called SIP/PQT/254202020

Is that a valid telephone number?
Assuming it is not NANPA, then that means it is country code 254, Kenya and the number being dialed is 202020.

Is this how your SIP trunk provider is expecting the call to be presented?

We sell Carrier Services and usually size based more on the number of calls not the number of people. 30 simultaenous calls should be easy on any good (read: true business class) Internet circuit. @scottalanmiller has the bandwidth nailed on the head above. We di try and get users to get the best Upstream bandwidth which for cable modem installs around here is very difficult (Time Warner only offers 5Meg UP even in Business Class which is pathetic).
For our needs internally we needed well over 100 simultaneous call paths. We went with a dedicated circuit to a Rochester based cough carrier cough and a backup trunk over an Internet circuit. While bandwith has never been an issue lets just say I can't wait for the contract for the cough carrier to expire so that we can port and go to the Internet for our needs and dump them!

@scottalanmiller said in FreePBX External/Remote Extensions:

@JaredBusch said in FreePBX External/Remote Extensions:

@RamblingBiped said in FreePBX External/Remote Extensions:

Option #1 up and running. Now to work on the OpenVPN part of the equation... 🙂

I have set this up but then ran into the problem of the OpenVPN port being blocked. Slightly hard to change once out in the wild.

On which end?

The remote end obviously. As I have full control of the server side.

@RamblingBiped said in Podcast: Defending Against SIP Hackers:

Wow, this is kind of right along the lines of what I've been researching on remote extensions...

One of the things that we routinely do is to block all non-free calling so even if our extension, PBX or platform were to be hacked, even if they gained access to our SIP trunk, they can't rack up long distance calls or whatever.

@JaredBusch I really wish I were going 😞