Hello!
I have conference number with [ru] sounds. If i calling it playing:
Playing 'vm-rec-name.slin' (language 'ru')
But if i invite external from fop2 it plays on english language:
Playing 'vm-rec-name.gsm' (language 'en')
In sip_additional_custom.conf ru language is enabled (language=ru)
And if i invite external from fop2 key # on phone is not working (if i want rec name, for example)
Hi,
Once fop2 originates a call, control is then done by asterisk. If dtmf does not work is not something we can fix, not sure why dtmf will cease to work when originating calls from AMI.. it does not happen to me at least. It could be a bug in asterisk itself(?).
Regarding the language, it is also something kind of problematic. I know it is not a solution, but it is what I did myself in the past: replacing english sound files with my language files, as in many ocations I had prompts being play in english: sometimes in comedian mail, or other applications.
Call is originated, and if the channel driver is sip, the language set in sip.conf should be honored, but it seem it is not. There is a way to pass channel variables on originate commands in ami, but fop2 does not set any right now.
Thank you for answer.
In case with dtfm, i try with another version asterisk and it worked fine. I understand with lang sounds.
And i have another question:
If i call from fop2, from conference greeting will play at once. Is it possible to play it after a connection?
I think the greeting is played on "ANSWER". If you have analog lines that signal ANSWER before the real ANSWER, you will have that problem. But it is outside fop2..
Best regards,