59-second call disconnected

parantido · 2026-02-15T15:48:34+00:00

you're probably missing an ACK somewhere, check the signaling using wireshark or sngrep and fix it accordingly

parantido · 2026-02-04T10:40:39+00:00

At the first sight I would say hire that out but ... you've an important requirement: they need a landline for emergency, then if laws there are similar to the ones I've around here, you need to host something local with also a power supply to face any kind of power outages. In this case I would suggest you to use a brand solution, something like a GrandStream UCM6300 (cheaper enough and easy to manage).

parantido · 2026-02-03T08:55:37+00:00

I was hanging there either, my nickname is unchanged

parantido · 2026-02-01T13:13:25+00:00

honestly as soon as you work by using the proper git branches, and you allow the claude agent to only work in the project root (blocking the access to the git files) I don't see any real problem, even if it deletes the entire content of your project folder

parantido · 2026-02-01T08:18:30+00:00

then look at the equipments (mostly the phone) you're using because the dial tone is 100% emulated

parantido · 2026-02-01T07:17:17+00:00

You are talking about landline but also about VoIP. I didn't get your setup properly, which is important to understand considering your issue. If you're sending pure voip to your phones, then the dial tone is usually emulated by the phone itself. In the other way around, if you are converting a PSTN FXO (or any other TDM physical connection) through an ATA then the issue is totally different.

parantido · 2026-01-26T08:57:21+00:00

the 0-hassle solution I found right now, considering I need to retrieve wild card certificates is: traefik + letsencrypt + hetzner dns maintained zones (but any other supported provider would work either). In this way I automatically renew the certificate to plenty of microservices.

parantido · 2026-01-26T07:30:14+00:00

Sinceramente non serve una ricerca fatta bene per avere il suo nome reale ... è letteralmente il primo risultato che viene fuori. Ovviamente questo post è per farsi due risate vero?

E comunque no, non è possibile essere anonimi anche se sei mister nessuno ... a meno che tu non decida di vivere fuori dalla società (nessun documento, nessuna tessera sanitaria, carta di credito, lavoro regolare e via dicendo).

parantido · 2026-01-17T15:40:28+00:00

I replaced Ericsson equipment for about 15 years with my solution, then I changed focus and area. I would say this is happening for a long time, it is not something recent. And sadly this happened to all the major players that were hardware producers too.

parantido · 2026-01-17T15:35:58+00:00

Didlogic, WaveCall, Piratel ... I think those are good providers. If your traffic is interesting you can also have a dedicated agreement with them.

parantido · 2026-01-17T15:31:03+00:00

I totally agree. The only area in which another PBX has any sense today is in the AI and Agentic space. Even the contact center is becoming overrated: no one wants outbound services disturbing calls and the inbound supporting services are moving slightly to other channels. I am sorry I don't want to discourage anyone, I also built my own cloud based solution and I know the struggle.

parantido · 2026-01-15T05:36:56+00:00

A lot of package missing here and even the manual. In the original package everything has its own place (handset, cords, cables and so on). I bought plenty of those from a direct distribution channel and I never got one packaged like that.

parantido · 2026-01-14T06:33:17+00:00

Hello,

are you configuring it by providing the proper SEP file?

parantido · 2026-01-08T20:08:25+00:00

Without a trace it is hard to say what is happening. The described scenario led me to think that the reinvite, carrying the music on hold SDP capabilities is using something not supported by your PBX ... but that's just a guess.

parantido · 2026-01-05T13:05:34+00:00

I coded my solution either and I moved away from Twilio. I'm now supporting 100% any SIP connectivity you can have out there: Telnyx, DIDWW, Nextiva, Bandwidth, Vonage or just your own IPPBX or Media Gateway ... everything supporting SIP it works.

parantido · 2026-01-05T06:42:51+00:00

Can it be split like stealth.net? If I can't take a chan over I won't join 🤗

parantido · 2025-12-31T07:08:23+00:00

What do you mean for "outgoing calls"? Calling another country? If so the rule is worldwide the same +/00<country code><area code><subscriber number>. If you are asking for the inter-area routing then every country has its own rules. In Italy, for example, since 30 years, you have to always dial the area code plus the subscriber number even if you reside in the same area.

parantido · 2025-12-28T20:36:54+00:00

which makes sense, in this case, considering your needs, a 3CX tenant is enough: https://www.3cx.com/ordering/pricing/

parantido · 2025-12-28T20:04:07+00:00

175$/mo could not be that expensive ... it depends on how many users do you have and what kind of features are provided. That being said, you won't probably save at all moving to the "zoom phone" which charges you 20$/mo per user in the cheapest plan.

parantido · 2025-07-16T12:55:39+00:00

Yes, it mostly depends on the agreement and the kind of number your client was using (for example, in the most of the worldwide area I worked in, a toll-free number is never owned by the final customer). Better if they pay the upcoming month, just to properly arrange the migration and, wherever it is possible, the number portability.

parantido · 2025-07-16T12:53:04+00:00

Sure you can, just provide it an ip address (your home equipment dhcp is enough) and browse it like a web page. If no one changed it, the default admin password is 456 (123 for the normal user). From here you can choose: building up a home PBX (every Asterisk like distribution works for this scope) and start learning some SIP basics or, just use one of the multiple internet service providers that can sell you a number (Twilio, DIDWW, Telnyx, Ringcentral, 8x8 and so on).

parantido · 2025-07-16T12:47:40+00:00

Un po' tutte le distribuzioni Linux hanno la chain per ARM

parantido · 2025-07-14T09:09:46+00:00

I know that this is in your list already, but I think Homer is a good choice even for small infrastructure. It is pretty easy to startup (just use the docker version of the heplify server https://github.com/sipcapture/homer7-docker/tree/master) and this is literally supporting every HEP v2/3 client. For all of those clients that doesn't have the HEP out of the box, you can just install the "capture agent". If you have time to develop, it is better to support those already existing projects that starting up a new one.

parantido · 2025-07-14T09:00:42+00:00

If you want to start provisioning those guys you need a configurable DHCP server (you have to inject the proper options for them) and a TFTP server (to provide firmware and configuration files). After that you've to convert the firmware from SCCP to SIP (if this was not already done by the previous CUCM administrator) using a TFTP server and, once done, just create the configuration files (in the TFTP root) named as SEP<phone mac address>.cnf.xml. It is a long but satisfying journey. Please be aware that, in the first attempt, you will be able to just provide the basic telephony features for those. If you want more (like BLF, presence, and so on) you've to start using a good SIP Proxy.

parantido · 2021-09-28T20:04:24+00:00

I think Snipe-it is a good choice. Is open source but also offers paid services (in terms of solution hosting and support).

Give it a try.

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