Morten Tryfoss

> With SIP status 183, you are about the answer (SDP response) of the Callee? Yes. > Or stated differently: From the point of Asterisk, you have a transcoding situation...

I've look a bit more into this. However, I'm not quite sure how it's intended to work. What seems reasonable to me for a simple call through asterisk with AMR-WB...

Hello! I made a proof of concept of something like approach 1 above. This is dirty work, but it seems to do the trick. https://gist.github.com/mtryfoss/3f84daa607993367c029ce3976527303 In short: -After parsing fmtp...

This is the line Ericsson sends in the INVITE to us: a=fmtp:96 mode-set=0,2,4,7;mode-change-period=2;mode-change-capability=2;mode-change-neighbor=1;max-red=0 It worked when I sent this in the INVITE back. I will try to narrow down.

It did not work with mode-change-capability=1, but worked well with mode-change-capability=2 (only this parameter). I'll check the Ericsson documentation for any settings related to this.

Here is the documentation regarding SDP from Ericsson. 


I understand. We are a small MVNO and the Ericsson device is our switch (MSC). We use a cluster of Asterisk-instances to add virtual switchboard services to our mobile subscribers....

I will be happy to help with testing and/or providing necessary development environment. Don't hesitate contacting me. Thanks for all the nice work so far!

I completely forgot this case. - .mode_change_capability = 0, /\* not supported */ - .mode_change_capability = 2, /\* not supported */ seemed to do the trick. The call was accepted...