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.
As a long-time lurker, I wanted to plug my company in case it is a good fit for anyone working on Telecom/VOIP for their company and may be evaluating new vendors.
THINQ is a software company operating in the telecom space. For outbound/Termination, our software gives direct access to 35 carriers without you having to negotiate individual contracts with them. You get to piggyback on our buying power of billions of minutes per month. For inbound/Origination, we have nationwide coverage at good rates: DIDs are $0.20/month and usage is $0.0025/minute
For Toll Free, we are the first to create a Toll Free Least Cost Routing software. What this does is analyze your traffic and update the routing templates with SOMOS on a WEEKLY basis to make sure your inbound Toll Free traffic is routed in the most cost effective way possible. Our average rate is around $0.0065/minute.
We also have APIs for origination (number ordering and provisioning) and SMS/MMS enabled numbers.
THINQ has no monthly minimum commitments to worry about hitting either, so if you have 10 numbers or 10,000 numbers, we may be a fit.
Message me and I will share my contact info and get you additional information and can setup a test account to play with.
We have an issue at one location where the customer hears 2-4 rings before the phone begins ringing on hunt group in the branch. When testing I can recreate the issue. Also while testing calls do not appear in the Trunk test tool until 1-1.5 rings The phone lines coming in are POTS lines. The carrier is the local iLec, Windstream. The switch is a ShoreGear 90 switch. I should also mention somewhere around ring 1-2 a click is heard and the ring is more like a half to 3/4 ring. When I hang up during testing I can hear the hunt group continue to ring for 2-3 rings after I hang up. If my onsite tester picks up the phone at that point it is dead silence.
Is this a known issue? Has anybody else experienced this?
I have recently bought a cisco 7911G and flashed the SIP firmware (SIP11.9-4-2SR2-2S) to it. I'm having some difficulty configuring it to connect to the SIP provider.
With this phone am I able to set it to directly connect to the sip provider (sip1.exetel.com.au) or do I have to setup a server with a PBX style setup like asterisk or similar?
I searched voip-info.org and created the SEPXXXXXXXXXX.cnf.xml file as the guide instructed. Have I got anything wrong?
edit: gonna upload the config file and link in a second, gets mangled in reddit http://textuploader.com/dfbcb
Hello folks, I'm a jack of some trades who tried to shy away from anything telco for as long as possible. Unfortunately it caught up with me.
I'm trying to help a colleague figure out the setup for the following VOIP configuration:
We have a customer with multiple businesses and separate lines at the same location with an upcoming switch from ISDN to VOIP. The provider won't/can't split the bills and keep the lines together, so they're going to set up 4 different WAN connections with their own routers...
We have a Panasonic NS1000 and a mediocre FW(not managed by us) onsite. As far as I understand, there is no way with the current equipment to split up the traffic because it resolves to the same provider.
Based on my googling I could proxy and differentiate the traffic based on the SIP logins with a SBC. Is that correct? If so, could you point me to a capable device? Or is this something I have to escalate at the provider so that they merge the connections on their end?
Cheers in advance.
I'm looking for recommendations on low-cost UC platforms. I was mainly looking at Sangoma and 3CX, but Sangoma's platform doesn't have mobile apps in production yet and I've heard a lot of negative feedback on 3CX regarding their attitudes, emotional management, etc., though an otherwise good product.
I've been using Asterisk since pre-1.0. I have hundreds of handsets off of various FreePBX installs, but I'm looking to work my way to a better user experience. That requires mobile apps, third-party integrations, etc.
Hey guys so im pretty new to this sub but I was wondering if I could ask for some help. I'd like to configure a SV9100 with SIP Trunk and given that I'm pretty new to VOIP as I've only dwelt on routers, firewalls, and access points administration. What are the things I need to ask my SIP trunk provider before setting it up? And if you guys have guides that I could use, feel free to post. Thanks in advance!
I am a volunteer with a small local non-profit. We have a phone tree service that we pay $120 a month to maintain, and it's pretty unfriendly. Can someone help us out by suggesting a better service?
Part of that payment is for service to have the hunt group reprogrammed all the time. The use case scenario is: our volunteers take turns receiving calls from our 24/7 hotline. We have our volunteers sign up to be 'on call' for a week to respond to the hotline. When someone calls, it goes through the hunt group until someone answers, if no one does, it goes to voice mail.
I'd like to have a service that does this, but lets us program in people through some web interface. Additionally, I'd love a pre-recorded message in a phone tree - something we do NOT have currently. Something like our meeting times and hours, then press 2 or something to reach a real person (that goes to the hunt group).
Is this possible? I'd LOVE to get more functionality for less price. Thanks, r/voip!
I'm planning to port my long-held landline number to Google Voice, and one of the steps is first porting it to a prepaid carrier. All of the tutorials I've read mention nothing about actually having to add minutes to your prepaid account after activating it, only that you want to port it off the prepaid as soon as possible to avoid losing your account and thus your number. Someone on the T-mobile site seemed to believe that I'd need to add at least a $10 time card to complete the process. Does anyone know if this is accurate? If not, how long will the account remain active before I lose the number, particularly with T-mobile?
Not sure if this is a question for the networking sub, but it seems more VOIP related... or at least in the neighborhood.
I have a Cisco SPA303 at the house connected to an external FreePBX server which uses voip.ms as its provider.
My home router is reporting a lot of this:
[LAN access from remote] from 5.62.63.x:2659 to 10.x.x.x:5060 Friday, May 18,2018 11:15:20 [LAN access from remote] from 192.111.139.x:25989 to 10.x.x.x:5060 Friday, May 18,2018 11:14:46
I can't figure out how these IPs are connecting thru the firewall to the phone. These IPs (as far as I can tell) aren't part of voip.ms's network.
I don't have port 5060 NAT'd on the router, so there's no way to connect to the phone from the internet; so I have to assume the phone is initiating the conversation, but I don't know how.
Asterisk is something Ive always wanted to setup at home but I believe after talking to coworkers Im very confused about all the moving parts and requirements as I thought a 56K RJ-11 modem was needed while I see people here with Pi....Very confusing..
First off, my current situation is ADSL and we only have one phone line; The current phones are three Siemens Gigaset C1, one for each floor. To explain things as simple as possible, the first floor is connected to POTS. When a call comes in, all three phones ring. First one to pick up, get the call and can talk normally. We can also call each other between different floors and obviously if someone picks up on the first floor, then can pass the call to someone on the third floor.
I understand that its highly possible in order to use Asterisk that I might want to change to IP phones. That would be a possibility I would be open to.
So could you please start guiding me, more on hardware? What do I need? What should I look for?
As of very recently I've replaced my companies old NEC PBX with a Switchvox system from Digium. We replaced a lot of the phones with Digium phones, but kept some of our analog lines including an analog call button (a Viking w-2000a). The vendor we worked through provided us with Digium SIP gateways to convert our analog lines, but I'm stuck on the call button. In the Digium gateway there is an option to auto-dial when the line picks up, but the call button appears to keep the line active at all times and send ring pulses when the call button is pressed. So the issue ends up being when I plug the call button into the gateway the gateway immediately dials out the extension and does this always no matter what. I'm not an expert on telephony but I do know that the call button was hooked into some sort of trunk on the old PBX that would push the call button calls to an extension as directed. Is anyone familiar with the digium gateways enough to make a suggestion or does anyone know of a cheap adapter that could create a live wire to the gateway when receiving ringing pulses? I would like to find a way to use this call button as we just purchased it as a replacement only a few months ago, but I'm also looking for suggestions on reasonably priced SIP based call buttons, anything that can work with just a single switched pair of wires would be amazing. If there is a better subreddit to ask I'd appreciate that knowledge as well.
TL;DR - Need to find a way to make an analog call button work with a switchvox system or looking for suggestions on SIP based call buttons that aren't super expensive.
SOLUTION: I enabled SIP ALG on my router and it worked just fine.
Like the title says, I have a strange audio issue on my Twilio/Asterisk VOIP setup.
Phone calls work great when calls are placed from my softphone.
But inbound calls don't.
If I call in, it rings, but when I pick up, nothing happens. If I hang up on either line, the other "stays connected" (is that a runaway call?)
No audio, not even 1 way, on inbound calls.
Again, outbound works fine.
I'm not sure how to troubleshoot this one.
Signalling seems to be fine, I can connect two lines, but the media stream doesn't seem to be working.
Network is a typical home set up. Modem -> router -> device. Ports forwarded, Asterisk server is on a static internal IP. My network settings within FreePBX seem to be correct.
Any ideas on where to start tearing this apart? I've only found threads on 1 way audio, really.
I found this but it's a little over my head.
I'm using one of the Betamax/Dellmont providers, but calling hasn't been working for the last 2 weeks. I have credit, I can connect successfully to the service, but if I start a call, it looks like it's making an outgoing call but all I hear is silence, the usual calling sound is not there. Also, the recipient's phone is not ringing.
Anyone experience similar problems?
In rural Canada, finally being able to move from ADSL to cable internet means that I am able to cancel my POTS line and go fully VoIP for my local number needs.
I have been using an Asterisk box for years with an FXS/FXO/ATA setup and an old Toshiba analog PBX.
Does anyone have a recommendation for a provider that will allow me to port my existing number, allow incoming and outgoing calls, has caller id and isn't very expensive?
I've been looking around for a few days now, but most places have terrible webpages that are confusing or just look horribly unprofessional, so I don't know what's legit and what's fly-by-night. Prices also vary wildly seemingly without rhyme or reason.
So I was having trouble provisioning my phone. I spoke to Vonage, and their tech support got me up and running by setting up a static IP and changing the DNS addresses. Now I can't use the IP address to log in to the config. Can anyone give me some guidance?
Quick question, hopefully this is the right place to ask.
I travel a lot and have all my 2FA stuff over Google Authenticator and a few other apps. But the 2 banks I am with do not support this. So I would like to have a dedicated VOIP number for receiving 2FA SMS's from the banks which I can access anywhere in the world.
I have considered using a Skype number and securing it with an App like Authy or something, but a Skype number costs ~$50 a year. Are there any alternatives? I can't set up Google Voice outside of the US (which I am right now), and, if I'm not mistaken, is only accessible from within the US. So I'd need a VPN for that, which is another cost.
Any advice would be greatly appreciated! Thanks!
Skype numbers do not receive text messages :(
I am probably going to get one of these two phones for a small office. Has anyone used both of them? Aside from the larger display and more buttons are there are reasons to go with the 2170? I like the smaller size of the 2135, and it seems to have what is needed. Is the sound quality the same on these?
I have a one person biz that I use Skype with. I want to dump Skype and find another solution. I need:
I will probably move to Mexico by the end of the year. I want to keep my current US based phone # I've used for 9 years. I do remote PC support for homes and small businesses.
In an obscure post on its support forum, Google has quietly announced that it will discontinue Google Voice support for XMPP on June 18 replacing it with a proprietary service tied exclusively to Obihai commercial hardware. This change will wipe out the communications capabilities of millions of unsuspecting users unless they purchase and deploy Obihai hardware before June 18. Details: http://nerdvittles.com/?p=25787
I'm looking for an inexpensive VOIP provider just so I can mess around with VOIP and get a feel for it. My main two immediate uses would be to easily and inexpensively change my phone number if I wanted to set up a new tinder account. As well as to eventually try to set up a chatbot like lenny on a raspberry pi box connected to my wifi to mess with spammers.
Does anyone have any recommendations to get me started? I'm looking for a cheap provider. Also what client side software should I use? What do you think about microsip?