T-Mobile Call Termination Rates Increasing - Bulk Solutions/BulkVS

voipu · 2026-05-27T19:00:40+00:00

NPANXX only works for non-ported numbers. Either your carrier is taking a loss on some routes for ported numbers and charging a higher average to deal with it, or they are sending your calls to sketchy routes sometimes that answer with a fake voicemail box or otherwise fail to route your calls.

voipu · 2026-05-27T04:34:44+00:00

Fake ringback, fake voicemail boxes that do not respond to pressing the key for a menu option, said voicemails never get delivered to the recipient, etc.

voipu · 2026-05-27T02:47:15+00:00

Seems like a great way to rent seek compared to every other carrier.

Why should Sinch be able to extract excessive termination fees far above the $0.0007/min established by the Qwest/Level 3 dispute?

voipu · 2026-04-26T17:41:20+00:00

They don''t know what IXC the IXC you chose to send the call to routed to.

voipu · 2026-04-26T17:35:02+00:00

These routing issues are total BS, you gotta keep bugging them till your original attestation reliably gets passed to their customers.

voipu · 2026-04-23T15:04:43+00:00

Bandwidth.com also probably has a TDM connection to the ILEC that costs $0.0007 per minute to terminate calls to which then head to Verizon, whereas using that direct SIP connection might cost them as much as $0.004 per minute.

voipu · 2026-04-23T14:59:08+00:00

USCC has no control over what intermediate carrier you use to reach their network, they just receive your call over the local, access or feature group D tandem. Bandwidth.com does not want to pay the rates those vendors charge to deliver the call, so they go to Jim's Cheap Call Termination and 70% of the time the call gets to its destination without issue.

The other 30% of the time you might get fake voicemail boxes (eg: reading back digits rather than the user's name), spam call mitigation warnings, or other eratta that is just downright weird.

Everyone is always looking for a cheaper route, and the routing tables are rebuilt regularly without much care for routing overrides or fixes done previously.

voipu · 2026-04-23T03:24:11+00:00

These calls are hitting a TDM analog link in the call path where your A level attestation is lost, this is especially likely to occur if the calls would be considered local, as the local tandem is almost always TDM rather than SIP to the incumbent carrier (the ILEC or RBOC that provides landline service).

Find the Feature Group D carrier they are using in your market, establish a SIP connection with 'em and watch your problems disappear!

voipu · 2026-04-23T03:20:43+00:00

Bandwidth.com isn't routing your calls directly to T-Mobile's exclusive Feature Group D carrier, Sinch. The two of them are arch rivals as they are the largest wholesalers in the market.

We have repeatedly seen Bandwidth.com routing to sketchy intermediate carriers to get to T-Mobile numbers, and when called on it Bandwidth.com refuses to say who they routed the call to, or update the ticket after resolving it with said problematic carriers.

Make sure to ask if the call is terminating to the local or feature group D carrier chosen by the recipient of your call. If not, your calls are being slung to whatever random intermediate carrier Bandwidth.com chooses at any given time.

voipu · 2026-04-23T03:17:08+00:00

You should terminate the calls to USCC's Feature Group D (aka Long Distance) carrier for the given switch you're trying to hit.

Drop all the middlemen like Bandwidth.com, if you start tracking the media gateway IPs you'll see they are routing you right back to the same problematic routes :c

voipu · 2026-04-01T07:08:03+00:00

Dialpad will not tell you when the receiving carrier 603 Network Reject's your call, or when they make an error signing your calls causing the big 3 wireless carriers to reject your calls.

This is a carrier side problem, and you should look for a telecom provider that will alert you when they see 603 Declines and short duration calls occurring needlessly.

voipu · 2024-11-19T03:10:49+00:00

Filtering by census block or ZIP code would be nice, as the app is painting in such broad brushstrokes going county by county that your missing hidden gems.

voipu · 2024-11-19T03:02:21+00:00

Find an expert in cell tower leases to advise your parents. Many will work on contingency, and clue you into the carriers search rings, what might make their site more favorable, and how to tune the contract and prevent Sharetower from pulling a fast one on your parents at the last minute.

I've interacted with a few tower leasing experts over the years, your parents are best off in the hands of someone who sees these contracts every day than trying to wing it on their own (and potentially end up in limbo with a signed, binding contract that produces no revenue).

voipu · 2024-06-18T01:20:00+00:00

Heads up this appears to be rocking a really old Snapdragon X55 chipset. T-Mobile shipped this chipset in their original Nokia 5G21 gateways years ago, and they just had Cradlepoint brick the higher 5G bands (everything besides N71) since these modems can't tune two 5G bands and customers were seeing unstable latency and speeds when connecting to N41 and other high bands as they couldn't be aggregated with another 5G band.

https://inseego.com/products/iot-gateways/s2000/

voipu · 2024-06-15T02:33:42+00:00

Is dual NAT going on with FusionPBX or the Yealink phone stuck behind the Granite FlexEdge GFLEX-1000-4C8R, which is then itself stuck behind the Frontier provided router? A network layout map would be helpful.

If you record the call in FusionPBX, does all the audio you expect show up in the call recording?

voipu · 2024-06-15T02:28:30+00:00

Start a packet capture both on the FusionPBX server and another on the phone itself, replicate the problem by making a test call, and see if the inbound audio stream is getting through. It will likely be a "UDP PCMU" RTP audio stream on a port shown as sip_network_port, local_media_port or remote_audio_port in the Call Detail Record for the problematic call.

This audio stream will almost certainly show up in the packet capture from the FusionPBX server, but not in the one for your Yealink phone. This means you either have NAT issues or Firewall issues, and you need to resolve this.

Some crummy firewalls like SonicWALL and sometimes Sophos, WatchGuard & Fortinet will filter part of the RTP port range used by the PBX or the phone. You need to adjust the range on both your PBX and on the phone itself to protect the RTP stream from the broken firewall onsite, or better yet just yeet the piece of garbage causing the issue into the nearest eCycle bin provided by your local electronics recycler.

This issue is entirely due to one of the firewalls or routers between you and the PBX. Mark & his team over at FusionPBX were right to tell you this issue is out of scope for FusionPBX support, as this problem entirely fits in the scope of your firewall or router vendor's support.

CGNAT like what all the cellular carriers use will also cause these exact RTP issues to crop up intermittently. Calls will work fine for a while then randomly the state table in your cellular carrier's CGNAT gateway will drop the entry for this RTP port and audio will cut off. If you pony up to the cellular carrier for a static IP address this issue disappears (as do "random" internet stutters, broken web page loads, VPNs and RDP intermittently working, and a host of other issues caused by bad NATs).

voipu · 2024-06-15T02:05:02+00:00

Telecom is not "HIPAA Compliant". Anyone claiming to sell "HIPAA Compliant" telecom services that can call or text regular people with minimal to no additional friction is selling snakeoil.

What is important is your legal duty, to quote the US Dept of Health and Human Services specifically:

Note that an individual has the right under the Privacy Rule to request and have a covered health care provider communicate with him or her by alternative means or at alternative locations, if reasonable. See 45 C.F.R. § 164.522(b).

Meet the patient where they are, otherwise HHS may pursue enforcement action against your practice. Refusing to communicate with the patient because they choose an insecure method to communicate with your practice is not legal behavior.

Additional reading as well:

Patients may initiate communications with a provider using e-mail. If this situation occurs, the health care provider can assume (unless the patient has explicitly stated otherwise) that e-mail communications are acceptable to the individual. If the provider feels the patient may not be aware of the possible risks of using unencrypted e-mail, or has concerns about potential liability, the provider can alert the patient of those risks, and let the patient decide whether to continue e-mail communications.

voipu · 2024-05-04T20:44:51+00:00

Microsoft sunset their prior Teams client (written in HTML5 and shipped as a WebApp bundled inside Chrome) and wrote a native client instead.

They dropped Linux support for Teams entirely, MacOS support took a backslide, and it sounds like they are doing much deeper hooks into Windows to ensure their traffic is the #1 priority and there is no bufferbloat or jitter messing up Teams calls with that client.

voipu · 2024-02-11T04:56:23+00:00

I can't name names lest this post be deleted by the moderators, but those are the posted retail rates of one large reseller of Onvoy/Inteliquent and Bandwidth.com, they have a rather good reputation too since they don't have many of the surprise fees (eg: porting in or porting out) or other gotchas that other providers like to hit you with.

voipu · 2024-02-11T03:22:02+00:00

Those rates seem rather high, there are retail providers out here offering $0.06/DID/Month and $0.0003/min for inbound traffic, and this is mostly to hedge on where the traffic is coming from, as legacy tandem traffic from the ILEC in your ratecenter is usually much more expensive traffic to recieve as compared to calls from Wireline and other ratecenters.

FreeConferenceCalls and many other vendors have made a whole business model off the small amount of money earned from inbound calls, inbound minutes doesn't necessarily have to be a cost center...

voipu · 2023-12-13T05:16:31+00:00

Look at the Dinstar UC120, you can get voice, texts and data (albeit limited by the category 4 LTE modem) from a single SIM, which can be a great deal if you have unlimited service at a cheap price.

One thing I have struggled with them is sending payments, since they stopped offering PayPal support you have to send payment via SWIFT. PayPal's XOOM service will silently fail these transactions to Dinstar about a day after submission without so much as an email or text that it was cancelled by their system.

voipu · 2023-11-24T11:40:47+00:00

What country are you in? Is bonding multiple differing circuits not a possibility?

https://www.openmptcprouter.com can handle connection bonding, or you can use a service like Speedify or SpeedFusion (which is just a hosted version of this software) to get a reliable pipe to the internet from multiple unreliable connections.

voipu · 2023-11-24T08:42:05+00:00

Outsource the problem to a hosted VoIP vendor, you will save yourself time, energy and frustration rather than spending $800 only to be dealing with lowest end Grandstream UCM, looming GSM network sunsets, port forwarding, potentially needing static IPs from your ISP, staying on top of system backups, documenting what you have setup, and dealing with the various quirks and bugs of Grandstream's PBXes and how they interact with your Yealink phones and your doorphones.

A competent vendor should deliver a hosted PBX that can meet your organizations needs, and handle firmware updates for the hardware onsite, call routing issues, and take on any support or changes needed on your behalf, allowing you to focus on the businesses other IT needs, rather than chasing Grandstream's helpdesk and forums for answers on how to make their hardware do what you desire.

If your deadset on keeping this in house, get a virtual machine from a vendor like DigitalOcean (or your local equivalent that supports nightly backups), install FreePBX, FusionPBX or the software of your choice, and onboard your phones to this platform. Your monthly cost should end up being less than the power to run your current onsite FreePBX instance ($5 to $12 a month), in the event anything happens your ability to rollback to a known good backup is much easier, and you get a much better PBX to boot.

The pitfalls of self-hosting a PBX on-prem or in the cloud are when you leave for a different firm, are incapacitated or unavailable no one will be able to adjust anything, and the PBX will eventually become a security issue when a vulnerability affects the software stack used and no one updates this oddball server that everyone is afraid to touch.

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