So some backstory. I am familiar with the issue in which a sip video phone cannot call via a sip trunk to the pstn due to the video attributes being put in the sip invite. The workaround has typically been to create a voice class on the dialpeer:
voice class sip-profiles 1
request INVITE sdp-header Video-Attribute remove
request INVITE sdp-header Video-Bandwidth-Info remove
request INVITE sdp-header Video-Media modify "m=video.*" ""
request INVITE sdp-header Video-Connection-Info remove
!
in recent ios releases I am encountering this issue with what I deem internal calls. That is to CUE (voicemail), and other sip endpoints on the system that are not video capable.
a video enabled sip phone can receive calls from all parties, but if it initiates them the calls hang and eventually result in a fast busy to any endpoint not video capable. I was able to add the voice class to the CUE/Voicemail dialpeer and that fixed the issue. However, I am unsure how I can trim the video attributes off of a call to another extension on the system. I assume this should be negotiated on its own but its failing to do so.
i am seeing this on 17.3.5 as well as 17.8.1 on a 4k isr. unfortunately I do not have a contract so I cannot open a tac case.
