Is anyone aware of any documentation related to asterisk or FreePBX that clarifies these fields for FreePBX or Asterisk?īasically curious if those relate to any settings or if those defaults elsewhere should be noted / used if we see anything else on any other equipment, etc. Many Asterisk / FreePBX notes / articles talk about “SIP INFO” - but not many of them seem to indicate anything about the INFO Type (DTMF-Relay / DTMF / Telephone-Event) or the DTMF Payload Type (which defaults to 101).
if the extension was being used in some type of trunking scenario) ? Basically trying to understand what I’m seeing here.ĭoes that mean the extension setting is ineffective? I’ve seen a LOT of people obsess over that setting over the years - OR does it mean that setting controlls what generation method would be used on that extensions tones were being generated on teh Asterisk side to send back to the device (i.e. So it worked - should it have worked? I have no RTP packets so there’s no way RFC2833 was somehow “leaking” through in the RTP stream. What I observed, is that Asterisk still sees the SIP INFO even when the extension is set to RFC2833. I did NOT change my extension DTMF Signalling on the extension - it is still RFC2833. I changed my DTMF on a Yealink phone from RFC2833 to be SIP INFO. But thus it is not possible to forward the CID in the forwarded calls, when using such providers in the outbound trunks.Han an interesting scenario which was causing me some RTP issues - so while I was having those issues, I tried a couple things. This is line of principle is normal, because the Dellmont providers wants to avoid spoofing the caller id. If the caller id is not in the authorized list, the call is received as anonymous by the PSTN phone. This works however the problem is that the Dellmont provider allows forwarding the caller id only if it is one of the CID’s that are registered in the Dellmont account. In the outbound trunk I have written the SIP credentials of the VOIP provider that gives me access to the PSTN. In the extension I have set the PSTN number to which calls are forwarded in the follow-me list. I have configured an extension, an inbound route, an outbound route and a trunk in order to forward to a PSTN number the incoming calls to a DID number.
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