In larger telco operations, SS6 and/or SS7 are utilized. You either send the tone “map” via a SIP INFO or SIP NOTIFY message. Out-of-band is literally done out of RTP. event “packets” in the RTP stream (see RFC 4733 and the deprecated RFC 2833)). However some implementations do it slightly different than others (ie. Today, we have a few different DTMF signaling methods. But not its analog perspective of it but rather how its (mis)implemented on the digital world. “I thought pressing 0 would go to the main menu!” There are many issues other than this too, however this is just the beginning of many problems a PBX trunk operator can face. works.™️ It has its own quirks as well, but I'll go into that a bit later.
3CX worked at the time I tested (around mid 2019), however I really don't like commercial “licensing” of products utilizing SIP.įreeSWITCH from SignalWire on the other hand. Kamailio kinda does, but breaks very often due to the VoIP config I mentioned above. SIP servers/gateways never completely follow the actual standards guiding the protocol.įor example Türk Telekom's “SIP server” is actually an IMS (IP Multimedia Subsystem) gateway from ZTE and sends incoming calls with the tel: URI instead of the usual sip: URI.Īsterisk doesn't support tel: URIs at all. However in the process of doing that, I learned how cursed SIP services are and even got a free Cisco IP Phone from an ISP contact. This led to many attempts to create a large home phone system. (I started writing this at the beginning of May 2020, and probably finalizing near July so yeah be warned.)Įver since I released my writeup about the (in)security of Türk Telekom's home gateway management, the VoIP config I found stuck in my head.
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