[asterisk-users] Is there a way to make asterisk send a INVITE in-dialog to re-establish the audio

[Joshua C. Colp]

On Fri, May 24, 2019, at 9:47 AM, Dan Cropp wrote:
>  
> We are working with an Avaya switch.
> 
> 
> We send them a REFER. If the transfer is successful, everything is 
> great. If it fails (busy), they send an INVITE in-dialog with a media 
> attribute of inactive. After that, they send a 486 busy.
> 
> The problem is Avaya basically put the call on hold so audio is not active.
> 
> The Avaya rep is indicating we need to send in dialog invite to get the 
> call audio back? They are essentially saying they put the call on hold 
> because we told them to transfer and it’s our responsibility to take 
> the call off hold.
> 
> 
> Is there a way to do this?

I don't think there is. We provide the ability in PJSIP to do a session refresh[1] but there's no ability to set the stream state like that, so I'm not sure what we would specify in that scenario automatically.

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_PJSIP_SEND_SESSION_REFRESH

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org