asterisk users - May 2019 - Is there a way to make asterisk send a INVITE in-dialog to re-establish the audio

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?

Have a great day!
Dan
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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

Thank you Joshua


-----Original Message-----
From: asterisk-users <asterisk-users-bounces at lists.digium.com> On
Behalf Of Joshua C. Colp
Sent: Friday, May 24, 2019 9:53 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Is there a way to make asterisk send a INVITE
in-dialog to re-establish the audio

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

--
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