asterisk users - May 2019 - asterisk-users Digest, Vol 177, Issue 11

Joshua
Is there a way in PJSIP to send the audio between the parties always,
unless one of the parties is behind a NAT?
A session refresh would work.
That my only problem with PJSIP. This is routine in the old sip channel.

On Sat, May 25, 2019 at 1:03 PM <asterisk-users-request at
lists.digium.com>
wrote:
> Send asterisk-users mailing list submissions to
>         asterisk-users at lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
>         http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
>         asterisk-users-request at lists.digium.com
>
> You can reach the person managing the list at
>         asterisk-users-owner at lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>    1. Re: Is there a way to make asterisk send a INVITE in-dialog
>       to re-establish the audio (Dan Cropp)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Fri, 24 May 2019 17:02:56 +0000
> From: Dan Cropp <dan at amtelco.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>         <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
> Message-ID:
>         <dabd7263f5bb401f95b3375f70d8960e at
AM-Mail2012B.amtelco.com>
> Content-Type: text/plain; charset="utf-8"
>
> 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
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ------------------------------
>
> Subject: Digest Footer
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ------------------------------
>
> End of asterisk-users Digest, Vol 177, Issue 11
> ***********************************************
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20190525/cfa02f4b/attachment.html>

On Sat, May 25, 2019, at 2:34 PM, Saint Michael wrote:
> Joshua
> Is there a way in PJSIP to send the audio between the parties always, 
> unless one of the parties is behind a NAT?
> A session refresh would work.
> That my only problem with PJSIP. This is routine in the old sip channel.
Any such functionality would be documented on the wiki[1].

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

-- 
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