Is there a way to make IP shuffling work on asterisk?
I am using asterisk version 188.8.131.52.
When call from IP phone covers to voicemail, the rtp packets flow between pbx media resource and asterisk.
Aferwards pbx requests IP shuffling, then rpt flows directly from IP Phone to asterisk but asterisk keeps sending rpt back to pbx media resource instead of IP phone.
Does anyone know how to fix this?
This is not s support forum!
Reading “re-invite” for “ip shuffling”. (Specically re-invites to dierect media, also referred to as “external bridging” in Asterisk.)
Reading “RTP” for “rpt”.
That is eaarly sub-version of a version that hasn’t been supported for about two years.
There were problems in the handling of chains of re-invites which were subsequently fixed. They may well still be present in your obsolete version.
The end of life version of 1.6.2 was 184.108.40.206.