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:
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.
this ended up being a jumbo frame issue. jumbo frames were enabled on the router and the switch, for whatever reason this caused the negotiation issue. fortunately we didn't really need them so removing them resolved the issue.
What you could do instead of using a SIP profile to strip the video stuff from the SDP is to add this to your outbound dial peer towards your service provider "voice-class sip audio forced". This will as well strip the video related headers from the SDP.