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Posted byFreePBX Developer3 months ago
Stickied post

Unfortunately, when you go to wiki.freeswitch.org/anything it's redirecting you to freeswitch.com without any hints as to where to go, rather than (In my opinion) the correct place of https://freeswitch.org/confluence/display/FREESWITCH

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

Sorry 8-(

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I'm having a hell of a time trying to get a new remote location up and running with all their phones. Let me give you a quick run down own the problem. We have our cloud-hosted (Rackspace) Asterisk PBX which was worked wonderfully with our main office. We've branched out and have new location. We don't have direct control of their infrastructure but can access their firewalls and phones remotely. They use Polycom VVX 410 as their phones and also have X-Lite (softphone) on their computers. The softphones seem to work great and we really haven't had any problem with them.

However, I cannot for the life of me figure out how to get the VVX 410 to stay registered and connected to our Asterisk server. They use a Sonicwall TZ-200 that no longer has any sort of support with it through Sonicwall. I'm hoping someone out there has experience getting their Sonicwall to play nicely with a cloud PBX and Polycom phones. Any help would be greatly appreciated. If you have experience, please let me know and I can further explain the problem.

Thanks!

EDIT: Here is a screenshot of what the peers look like on the Asterisk server end. Some are reachable, some aren't. Most soft phones that are on are registered too...wtf! https://imgur.com/a/hyqDBZJ

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Wondering if any of you have a suggestion for a cordless phone with base that'll register and work with with Callmanager?

Preferably the phone base will have an ethernet port for registration and one for client data access.

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I use VVX400 series, 401s, 411s, etc. If I wanted to turn off group paging on one phone, do I not do that by logging into that specific phone and disabling group paging? I just left the office and they are saying because I disabled it on one phone, that it disabled it on all the phones. I just want to disable it on ONE specific phone.

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Does anybody have experience using these here? If so, how does this functionality actually work? How is it keeping track of extensions in its local site?

If the main PBX is an an off site data center and all routes from the site to the DC are degraded, can a Vega take over and route calls between extensions within the site and also failover to PRI for outbound calls if needed?

What are the options to handle local survivability with an Asterisk based PBX? The few results I can find regarding local survivability always talk about Cisco or Avaya systems...

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I didn't have much luck on /r/ios - I was hoping you guys might be able to help.. As per https://old.reddit.com/r/ios/comments/8z6t1w/linphone_alternative_for_receiving_sip_calls/

Hi, I've used until recently Linphone to receive incoming calls from my asterisk box. Unfortunately, sometime in the past month they blanked blocked all (most?) VPSes to reach their IPs. I've tried 3 different VPS providers and I can't reach any sip.linphone.org nor www.linphone.org but I can reach them from my residential IP.

Is there any alternative (ideally free or at least a once off low cost) I can use to receive incoming SIP calls to my iPhone?

A while back there were some chan_skype ways to forward SIP calls to Skype but it no longer seems to be the case. I have an office 365 subscription (and Skype for Business) but couldn't find a way to figure out if I'm able to do SIP->Skype for Business. Slack also could be an option (as they have a voice feature) but couldn't find any way to integrate with it.

I think Bria (or some other paid app) has a paid option but I only receive one or two calls per month so it's more cost effective to simply redirect the call straight to my mobile number and pay a few cents per min or so but it has the disadvantage of not knowing the caller id (as I get my own number as CID) and it also prevents me from having a data-only sim card in my phone (I rarely use voice/sms anyway)

I tried searching for this for quite a bit of time but couldn't find anything useful.

Thank you.

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I have 3 softphones working just using sip credentials on https://voip.ms. We're looking to transition to actual desk phones. Does anyone have advice for the best phone that can just be setup by entering a sip address and user/pass? Do phones that work over WiFi exist?

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I'm based in the UK but spend a bit of time in the US, and would like to be able to use services like Venmo that require a US phone number to receive SMS verification (Venmo also requires a US bank account, but I have a virtual one).

I've spent the last half hour looking for some sort of service or app that will get me a US number that routes SMS to an app on my phone (or anywhere, really), but I've not found anything that does what I want for a reasonable price. Can you guys recommend anything?

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Can anyone recommend a phone with GREAT echo cancellation?

The room is about 20'x10' with a 8' ceiling on one side and 15' on the other. The 8' side has a 4x12 window and the 15' side is brick, as is one end.

It would be an awesome reverb chamber for recording. It sucks for conference calls. Bad.

It used to be capped with a dropped ceiling and acoustic tiles and there was a smaller window with blinds. Against my recommendations, they remodeled into an acoustic nightmare. Looks nice. Sucks for its intended purpose.

We had a quote to have someone in to help with the acoustics, but they don't want to change anything in the room... LOL. fml.

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Hello All,

I currently manage a small doctor's office and we have ATT. We have multiple calls coming in at any given time and right now we have two lines. What we were wanting to do was have the main number ring, pick up, then a 2nd call come in, click over (call waiting) and then a 3rd call come in roll over to the 2nd line. This does not work so the 2nd call rolls over to the 2nd line and then a 3rd call goes to voicemail.

I was looking at Vonage to help with this issue because of pricing being one of the biggest factor and ease of use being the other. Any thoughts on this? I'm down in Houston and something local would work or even another company. As a customer, I would hate to call an office and then get rolled into a voicemail. Thanks!!

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We are running Incredible PBX 13 on a Pi3B+. Expecting Max 3 concurrent calls ever. (Current system maxes out at 2)

Is it possible to have an IVR that is presented on initial call that is basically "press 1 if you are a new user, press 2 if you are returning user, press 3 if you are a vendor" and then anyone who presses 2 or 3 never gets the IVR again? We just want to make sure new users get routed to a different voicemail than existing users if they can't reach a human, and would not want to bother existing users with the menu every single time they call in. It seems like this should be possible but I'm not sure where to start.

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I didn't come across any relevant hardware posts, so I wanted to create this thread and hopefully get led in the right direction.

At my company we are using the Polycom VVX 310, so far its been a great phone for most of the company. It pretty much nails the few needs we have.

  • POE
  • 100/1000 Ethernet
  • Intuitive

As we started to experience a surge of calls the CSR department's phones started to introduce a little lag, and unresponsiveness. So i'm led to believe that the physical components of the phone are dated or just not powerful enough for our needs.

So here in-lies the root of my quest,

  • Have any of you fellas experienced 'laggy' phones during slightly more active hours?
  • Are you aware of any devices that are a more performant with better specifications? and if so how did you find it?
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Since firmware 5.8 the VVX x50 series of phones now support the obihai wifi usb adapter. Has anyone used one yet?

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I am trying to find new ATA or phone hardware that supports TLS and SRTP and works well with Asterisk. I only needs 1 line.

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Looking to replace the Pi. It would nice if I didn't have to rebuild the config.

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Is anyone familiar with CallRail? https://www.callrail.com/

I've been looking for a VOIP provider that can handle the following:

Texting

Outgoing calls from Cell phone without using data (use cellular voice)

CallerID spoofing (for numbers I already own)

Custom call flow creations (i.e. if this then that)

- Option to ring one phone, then another after no answer, then another, etc

A good mobile app that integrates with my existing contacts

I ran across CallRail. It ticks almost all of those boxes except the mobile app doesn't integrate with my existing contacts. So I have to copy and paste phone numbers from my contact list into the app to make an outgoing call.

CallRail is really focused on Pay Per Click marketing. So it's designed around those type of features. They aren't marketing it as a general VOIP software for just making and receiving phone calls. We don't really care about tracking where calls are coming from specifically (clicks, radio ads, print ads, etc).

If anyone is familiar with CallRail, is there another VOIP provider out there who maybe uses the same back end with a better mobile app? It's likely that CallRail is not using someone else's back end, but it's possible I guess.

Thanks in advance.

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I currently have Nextiva and I hate it. The android app used to crash all the time and now it doesn't work at all, it seems many others are experiencing the same thing. Checking my voicemail is a pain to do through the phone. I had voicemails sent to my email which was much easier but I recently found out some voicemails were just not making it to my email. I missed a few customers before I realized, cost me a lot of money. When I called Nextiva after waiting on hold forever I got customer support that spoke such broken english we couldn't understand each other. After my 3rd call, I got a fluent English speaker and they told me they are aware of the email voicemail problem and are working on it.... great! Also, the general ease of use setting up and changing things, I find not great.

What I need in a new service:

I really don't need very many features, Nextiva has more than I need. I currently have a Polycom VV411.

I would rather (not absolutely necessary) BYOD easier and cheaper to get cordless phones since I don't always have someone near the corded phone and there is no cell service here.

I need to keep my current phone number.

I need to be able to easily check voicemails while out. Prefer visual voicemail, I find it much easier and faster.

I need to be able to answer and make phone calls from my cell phone while out.

If the phone is being used or the power goes out I need calls to go to voicemail.

Also, I'm not even sure if I need business VoIP, would residential work just as well and cover my needs?

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Hey guys,

Need some input, i need to create a small call center where we will be taking calls and sometimes patching them in.

This is inbound only call with periodic patching thru to the account holder.

I was researching ringcentral office solutions for 34 per line, and also integrating with Hubspot since hubspot is free(i know i will have to pay someone to customize it, but i'm okay with a one time cost)

The primary business goal is to take inbound calls for my clients, but want to CRM to pop up the client's account info on each call, and i want to be able to enter in stats of the call/notes.

At the end of the month, i just want call reporting/recording/ date of call/ time of call.

I'm just one guy bankrolling this, and would like to most cost effective solution, that i can scale if the company succeeds.

Am i in over my head?

Tahnks!

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What is the best VOIP residential home phone service?

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Anyone have experience setting this up? I've gone through their "OpenVPN_Feature_on_Yealink_IP_Phones" document, as well as the Administrator guide.

 

I'm trying to set up a Yealink W60P to tunnel via OpenVPN into my network and then register to my SIP server. I've tested the certificates and keys on both Windows and Linux and I know they work.

 

I've created the tar file with the proper file structure:

vpn.cnf
keys (folder)
-ca.crt
-client.crt
-client.key

 

In the vpn.cnf, I specified the static path as documented: /config/openvpn/keys/ for the certificates/keys.

I created the tar, uploaded it, and restarted the base station. It just fails to register.

 

I don't even see any traffic in the logs on the OpenVPN server indicating it's attempting to connect and failing. The logs on the base station show the SIP traffic attempts, but nothing relating to OpenVPN.

 

Any advice/suggestions?

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I am working on getting it setup and making calls. I thought I read somewhere that for the trial you can only use a Sangoma phone?

Is this correct? I figure I would just use a softphone to test and see how eveything works. I am having some issues with it not registering a device and when I look under extensions it only shows the brand Sangoma.

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I'm using Grandstream Wave because MizuDroid died on me out of nowhere. With Wave, they can hear me, but I can't hear anything from them.

Anyone have any ideas or advice?

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We have PRI service through Charter/Spectrum/Whatever they're calling themselves these days. Over 500 DIDs.

Current setup: CUCM -> Cisco 3825 T1 card -> Charter Adtran -> Charter Cisco ME3400 -> Charter Fiber

I know they're basically doing a SIP trunk between their fiber and Adtran, and handing that off as a T1 to us.

We use Variphy for call and capacity reporting. We've hit 23 simultaneous calls on our PRI for about a minute a few times over the last few months. Talked to our rep briefly about adding another PRI and bonding them together for more capacity. He suggested I think about setting up a SIP trunk instead of PRI.

What advantages would this bring us, besides what I'm assuming is flexible concurrent call capacity? How would this affect our faxing? Right now we have about 100 DIDs assigned to route over an H.323 gateway to our RightFax server...would SIP trunking screw with that horribly?

Thanks as always for any insight provided.

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I need a local number in Argentina (BsAs) in order to make inbound AND outbound calls.

Which provider would you recommend? We're using FreePBX if that matters.

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We are a small shop with only 1 person on the phones, and she's chatty, which is fine our clients love it and it builds good relationships. We are investigating PBX offerings right now, and our top choice is ring central.

The issue I am concerned with is we will not have someone constantly in front of a computer monitoring a call queue, so the call handler could be on the phone, having a topical conversation with someone while someone else is in Queue who needs assistance, get's annoyed after waiting for 3 minutes and hangs up.

Is there any way to get notification on the desk phone itself, be it a blinking light, or an audible tone like "call waiting" on pots, to alert the call handler there are other calls in queue?

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So after a lot of research and personal needs assessment, I decided to drop my POTS line and go pure IP.

I'm in Canada, wanted to port my existing rural number, wanted a SIP trunk to a self-hosted PBX, had very little actual use on the landline and wanted to keep my costs as LOW as possible.

In the end I went with DryVoIP.ca, and the service uptime, features, costs and support have been about as good as possible. I'm over a month in now, and feel confident in making a post about my experiences thus far.

I looked at a bunch of other options, all of which either didn't support porting my number, or only offered all-you-can-eat plans that were no cheaper than my telco.

I decided to rebuild my FreePBX instance into a simple no-gui Asterisk box on Debian. I received good support thru the r/DryVoIP beta test group, and then posted this minimal config there once everything was working.

Number porting took about a week and was no big deal.

Costs are amazing. $15 to port a number, $2.50/month per channel and $0.75 per month e911 fee on at least one number. Costs are all billed in Canadian dollars.

There also isn't a port out fee like some other providers seem to have.

The included features are exactly the ones I wanted: voicemail with email and speech to text transcription, spam rejection and an sms gateway web interface. All work exactly as advertised.

The DID routing interface is easy to use and did everything I wanted (admittedly simple)

SIP user editing is also simple, but doesn't allow locking registration to a single IP, only username/password. I have a static IP, so I would prefer the ability to use both, but it isn't a deal-breaker for me

They only allow Canada and USA long distance, which to me was a feature. If I need international LD at some point I'll use another 3rd party. This may be a minus for others.

The only feature they don't have that I really want is encryption (SIP TLS/SRTP), which I will be bugging them about in the forum (and totally willing to be a guinea pig for)

I don't know if there is much else to add. I had simple needs and DryVoIP.ca met them at a minimal cost with excellent service to boot.

Anyone else with experience with them? I'd be interested to hear other perspectives

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