DTMF mode issue

(Al3mary) #1

i’m using freepbx with Asterisk Version: 16.4.1
I have Sip trunks with service provider, I have issue to calling some numbers,
the reply of SIP trunk service provider " Your PBX is calling some VOLTE user, so when INVITE is sent from PBX with SDP param (a=fmtp:101 0-16), then PBX received response 183 Session Progress with SDP parameter a=fmtp:101 0-15 . Now as per standard RFC 4733 this SDP parameter must have values from 0 to 15 only. But PBX is sending values from 0 to 16"…

in trunk I’m already using dtmf mode = RFC 4733

How i can change in Invite message a=fmtp:101 0-16 to a=fmtp:101 0-15 to fit SP requirements ?

(Al3mary) #2

Can any one help me with this issue? :cry:

(Ted Mittelstaedt) #3

Your SIP provider is being a d-unprintable

Event 16 was used for hookflash that was defined in an earlier RFC. rfc4733 deprecated hookflash but everyone still supports it - or they ignore it - and a 0-16 is VERY common “in the wild” Asterisk sends it because there’s some phones that support it and some places that still use the hookflash for transfers. While it is certainly fine for a provider to not support hookflash since tons of systems out there send 0-16 in the INVITE your provider should have certainly seen this a hundred times before. It won’t kill them to just ignore the extra code.

You might explain to them the basic principle of a customer/vendor relationship, YOU pay THEM for doing it YOUR way not doing it THEIR way.

(system) closed #4

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