VoIP

Hey r/VOIP!

I'm looking into Sangoma SBC's for a potential future project, and was also looking to see what they have for training programs.

Regarding the SBC's, any one have anything to say about them? good or bad? Comparable to any of the big names like ribbon/sonus, metaswitch, etc, or closer to what can be had for free?

I know there's mountains of documentations and what not online, but I'll be completely honest, I'm a terrible self learner when it comes to things like that. I'd much rather go through a training program and get certified properly.

I was looking in the Sangoma portal and saw they have a 2 day course for SBC Getting Started Administrator which includes the certification exam

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Thanks!

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I am currently porting numbers away from our local telecom company. I have been doing it department by department as we have alot of DID's. The problem that I am worrying about is the local telecom has been firing people left and right. The writing is on the wall they are closing shop. What happens to the DID's that I have not moved yet. If they close how do I get them transferred over.

Need help connecting an IP phone. My computer connects wirelessly and I don't have a network port handy for the phone. Can I connect my Polycom WX250 via the computer's unused network port that is bridged to my wireless port? Windows 11. I want to avoid purchasing a wireless adapter for the phone.

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Many thanks in advance

Hi all,

I would like to hear from community what are some major issues or gotchas with Kamailio.
For me one major issue is just testing and confirming your configuration.
Also I found out with some loadtesting topos module really hits database.

Do you use redis with Kamailio? Any issues?

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It was announced last week that Ooma has bought 2600hz. One less CPaaS option in an already small pond. Thoughts?

Hi all.

UK user here. I've just converted my land line to digital and bought an SIP phone on Amazon - Yealink T46U.

I've plugged this phone into my router, it's detected the network fine, and all firmware has been downloaded and up to date.

I can hear a dial tone, but I don't seem to be able to make or recieve calls, so need to check the SIP phone settings. However the settings are password protected on both the phone and the web interface. Issue is: the default username/pword in the manual (admin/admin) isn't working! I'm tearing my hair out here - I've tried all combinations of capitals but to no avail. Access denied.

Two things occur to me:

1. Could it somehow be downloading username and password details for VOIP service from my service provider? (Zen Internet). Not sure why this would be, as I can plug a standard land line phone into the 'FON' port on my router and that all works fine.

2. Have I bought the wrong phone? Ie is this phone meant for an enterprise setup where there is a dedicated provisioning server involved and not for lowly home users like me? Originally I tried this with a Cisco phone (had it at home from a previous job where recorded lines were required) and I tried to plu the Cisco phone in, only to be asked to enter the Cisco server address. A quick search informed me that with Cisco phones you need a dedicated server in an enterprise environment. Could it be the same deal with this Yealink phone?

Any help on this gratefully appreciated. Thanks all!

I’m trying to set-up my home voip system. I was trying to use the Linksys PAP2, but after two failed eBay devices I don’t want to risk buying a third “new” unit and have another issue.

Can anyone recommend a ATA that is affordable and where I can purchase it new. I’m not interested in a all inclusive like ooma.

Thanks!

I've been using SIPHarmony for more than a year, and I just can't continue to use their mobile app with the lack of progress on fixes.

I'm looking basically to use my US based number anywhere that I have internet, as I travel the world pretty frequently. I'm looking for something with a mobile app, and SMS texting is a necessity. Number importation is important, and reliability when sending and receiving picture messages. I'll likely be moving 6+ lines. If there's a plan that allows you to call and text Filipino numbers similar to Vonage's Philippines 3000 plan, that would be awesome, but it's not strictly necessary.

Any recommendations?

I want to sign up with voip.ms and I'm wondering if there's a way I can just have my computer speakers ring when i get a call, and I answer with a simple headset/mic? Or else, can someone give me an idea what a minimal setup looks like? I'm trying to figure out what I will need to buy.

I’ve factory reset this phone and installed latest Firmware. In RC there is option of T57w phone but not this t58w. Inexplicably, there seems to be no way to manually add this phone.

Is it possible that this phone simply won’t work. The office has T46S out the ying yang and they work fine.

I bought a Grandstream HT813 to integrate my apartment intercom so it rings my VoIP phones internally and cell phone externally. The complex I live in has a different ring pattern for calls from the panel downstairs. It's basically 2 short bursts of ring.
I have a telephone line tester and it is confirmed to be working properly but this uses the regular c=2000/4000; ring. I've tried playing with these numbers. Such as c=1000/250-1000/4000; It still can't seem to pickup on it. It's similar to the UK ring but longer ring times.

I recorded an old phone ringing to this pattern when called for which I have the file if anyone is interested. Perhaps someone can give me an idea what the correct numbers should be.

The other thing is, why does it matter what the ring candace is. How about just pickup the phone anyway? or have an option for "ring candace learn" if it is required?
I mean my brother printer/fax can pick this up on the 2 long setting or simply answer it on fax only mode.

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At the risk of being told to RTFM or to do a search on the forum I wanted to make a quick post to ask for help to set up a very simple solution to link my USA-based clients with my Africa-based call center workers.

It is a very simple setup. Max 50-100 calls per day, maximum 10 call center workers (starting out with 1-3). I want my clients to be able to call my business number, which I've had for around 10 years and to be routed to the call center workers who can take their calls and answer questions or for me to be able to selectively pass them on to the call center workers after I've spoken with the client directly. I would also like the ability to monitor and record these calls (for quality assurance purposes, highly preferably with a notice/warning that occurs at the beginning of the call, as is legally required) both in real time and through a recording.

Would Zoom Pro Global Select achieve this? Are there better options? There are no countries in Africa that are listed on their Coverage List but I am not sure if that relates to the country of origin/business country or the country where the call center workers are located. Since it's VOIP I would assume that with a robust data connection that anyone from pretty much any country could connect and then the "Global Coverage List" (https://www.zoom.com/en/products/voip-phone/features/global-coverage/) specifically relates to the country/countries where clients are based/the number that the client calls to access the business, right?

Very appreciative for any and all help that anyone can offer, including better solutions. I'm looking for the simplest solution that includes the above features (namely recording the interactions) and I'm not necessarily super price sensitive but I definitely want the simplest fully functioning solution. Downloading an app, having the call center workers download an app on their phone/computer and then putting in some credentials and ready to go out of the box type of experience.

I have self-hosted kubernetes cluster, launched on oracle cloud instances.

I want to launch there FreePBX (asterisk distribution). So, I have problems with NAT - sip protocol is terrible to work with NAT.

How can I bypass it? I have two ideas:

1. create VPN (openvpn??) service in the another container in the same pod with freepbx, so, clients(app on smartphone, hardware phone) will connect to vpn, and than to freepbx
2. use turn/coturn, something like this: https://github.com/l7mp/stunner

So, client will use turn, and freepbx will use turn as well

But maybe I can just somehow configure SIP protocol for server?

Basically, I have multiple NATs: client > router > external IP > internal IP of oracle cloud > metallb (which uses this internal IP > pod's IP address.

And it kills SIP traffic. It even can't auth :(

Also, I used gomplate template to generate ports for my service (for SIP I need high range of ports for RTP)

Hi! I just came in possession of a Cisco 7942 IP phone and I am eager to set it up. I did some research and discovered how to flash it with SIP firmware as opposed to the SCCP firmware. As for the actual SIP server, I discovered Kamailio and decided on this one as it is in the main package list for OpenBSD which is the platform I mainly want to develop this on. I also discovered Siremis as a GUI for Kamailio, but it seems development has been lacking. Does anyone have any tips or tricks or advice on setting up a SIP server? Anything weird I need to look out for? Any challenges you guys specifically faced when setting up a SIP server? Thank you for any tips and advice!

Hi, I am starting my business from another country in the United States and to seek help, i have to explain the whole situation. So the thing is, we are gonna remotely connect with the screens of restaurants located in US. Customer is gonna call to place an order either for pickup/delivery. I need to know how can i configure phones from that restaurant to my country so that when somebody calls on that restaurant i would be recieving that call here.

Hello Reddit, could someone please advise me with any guidance on the following. I received harassment, threats & so forth via text messages from 5 phone numbers. I filed two police reports. It lasted approximately 5 days. It was beyond comprehension traumatic and I need to be my best advocate now. In the midst of the 5 day insanity I was approached by a private party who received messages with my information in them and accusing them of being me. We don't know what connects us and the individual has a private investigator and attorney working for them regarding the messages they received. I need the VolP data or guidance as to what I need to obtain information/ sourcing of those numbers etc. Coincidentally apparently this has happened to the other victim whom has the PI and attorney and it was found to be two ex girlfriends. One of those ex girlfriends apparently also recieved messages from the same 5 numbers. Both those exes engaged in such text warfare with their client throughout last 6 months to year and they were able to apparently stop it so the victim did not push charges or go to police. Any help or guidance would be great, thanks Reddit!

Hello, everyone! I work in a dispatch team for an US based trucking business and among the tools we have, we’ve been using Google Voice as an exclusive tool to communicate with our drivers, both text and calls. All of a sudden, Google Voice has suspended our account, though that didn’t really come as a surprise and we are fairly sure of the reason, since Google Voice only works in the US alone, along with some other states around the workd, list of which our country unfortunately is not a part of.

These being said, we are currently looking for a tool to replace Google Voice in our arsenal, as its simple interface and accessibility served its sole purpose of allowing us to keep in touch with our drivers extremely well, it was all we needed as far as that goes. I’ve started looking for stuff around and found TextNow, whose interface was as simple and easy as Google Voice’s and it would have been the perfect replacement, if only it wasn’t plagued by the same restriction GV has and that it can only be used inside the US.

Pleas let me know any suggestions that you might have for a free/paid VoIP that would fit those criteria, something that is simple, fast and smooth and which would allow us to keep in touch with our drivers much like Google Voice did.

Has anyone noticed issues with multipart messages to Telnyx? Even purely text based messages are having issues when they are more than 160 characters so they have to be sent as multipart. Anything under 160 is pretty solid.

It's not even that they all fail. Some 250 character messages arrive. Sometimes I will send it once and the text fails, I wait a few minutes and send the same text and it succeeds. The texts all show as sent from the carrier, and don't show up in the Telnyx logs at all.

It's also a very consistent issue to reproduce. We have tried from AT&T, Verizon, T-Mobile, and Twilio. Send a few large texts and 20-70 percent will fail to arrive.

Just wanted to see if others are seeing this, because I assume it is across all of Telnyx.

We have a couple of SNOM PA1+ units using the lineout to an external amp's aux port. Everything is working correctly, except the audio is really quiet (two big PA speakers you can only hear if you are within 20 feet of at max volume).

If I remove the aux cable from the line out of the Snom and place it in my phone it's really loud and clear. I've tried hooking headphones to the Snom's line out and they are medium volume.

The volume is maxed to 15. I don't see an option to increase vol_speaker in the settings. I appreciate any help.

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Snom PA1+ too quiet settings

Hello folks!

We've noticed a sudden influx of brand new accounts with 0 karma posting "reviews" of Nextiva that all read exactly the same.

Now, people endorsing a provider isn't typically suspicious, but six hot-off-the-press accounts posting glowing reviews of Nextiva (at the same time!) in every thread doesn't pass the sniff test.

Fortunately the karma/age filter blocked their comments and they have now been banned. However, those of you that these bots replied to will likely have a bunch of notifications. Nothing we can do about that, I'm afraid.

If these bots actually were commissioned by Nextiva, stop it. Trying to disguise your shitty infomercial-esque advertisements as legitimate reviews is scummy.

so. is there any free disposable phone number sites/apps which would bypass the "you can't use this phone number" thing when you try to type the number in a certain site, for example, discord asks you to type in your phone number to verify your account, twitter locks your account and asks you to type in your phone number, etc.

so yeah, is there any free disposable phone number site/app which would make them accept the number if i already have my own phone number verified in another twitter/discord account?

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Hi, does anyone use GoTo Connect and are in the UK?

I'm running FreePBX and a Grandstream HT802. It registers to FreePBX fine and have had no issues.

I've gone through the process of moving my desk phones to a different VLAN (20, from VLAN 1), which has a different IP subnet. So now the phones have a different IP and they successfully connect to FreePBX, which has a separate interface on that VLAN. So far, so good.

The problem is when I moved the HT802 to the new VLAN and IP. I won't connect to FreePBX. The Asterisk log shows nothing (as in, no attempts or registration errors), and Fail2Ban has no blacklisted IPs. I even whitelisted the IP for the Grandstream ATA to no avail.

Here's a brief of how the network configuration looks:

VLAN 1

FreePBX IP: 192.168.1.10

HT802 IP: 192.168.1.3

Status: Successfully registers

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VLAN 20

FreePBX IP: 10.10.20.10

HT802 IP: 10.10.20.3

Status: ATA says "NOT REGISTERED". FreePBX doesn't log any failed attempts, that I could find.

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The steps I performed on the ATA is to update the IP for the FreePBX server, click apply, then click update, then reboot. Just as I click REBOOT, I change the VLAN assignment on my Cisco managed switch to VLAN 20. It boots up to its new IP, but never registers.

I've tried setting the ATA to use DHCP, I've also set it to use Static IP... no change.

I change the IP for the FreePBX server to register to, to point back to the 192 address on VLAN 1, click apply, then update, then reboot. Immediately put it back on VLAN 1, and then it registers.

I've also done a FWCONSOLE RESTART on FreePBX each time I've moved the ATA to a different VLAN, to no avail. It simply won't register.

I really don't understand why the Grandstream device won't register. Oh, and it's also running the latest firmware (I did upgrade the firmware, but didn't change anything).

Anyone ever come across this before?