Hi, I have been testing Chrome JSSIP/Sipml5 with OSDial 3.0.2. I can get SIP calls to connect between chrome browsers but get no audio. This seems to point to a bug within asterisk around strp.
If a build a generic 11.19 using ./configure --with-crypto --with-ssl --with-srtp --with-uuid=/usr/include everything works perfectly.
I have a couple of questions:
Is there anytime line to get Osdial 3.0.2x working with asterisk 11.19+ or asterisk 12?
Could you provide a list of configure switches OSD was compiled with in-order to attempt to overlay asterisk 11.19 over the current OSD 11.13-54?
Do you know of any patching/fixes that can be applied to OSD 3.0.2 or its asterisk 11.13-54 that would make JSSIP/Sipml5 work correct within this installation?
Many Thanks,
Paul