<div dir="ltr"><div class="gmail_default" style="font-size:small">Joshua</div><div class="gmail_default" style="font-size:small">Is there a way in PJSIP to send the audio between the parties always, unless one of the parties is behind a NAT?</div><div class="gmail_default" style="font-size:small">A session refresh would work.</div><div class="gmail_default" style="font-size:small">That my only problem with PJSIP. This is routine in the old sip channel.</div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Sat, May 25, 2019 at 1:03 PM <<a href="mailto:asterisk-users-request@lists.digium.com">asterisk-users-request@lists.digium.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">Send asterisk-users mailing list submissions to<br>
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Today's Topics:<br>
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1. Re: Is there a way to make asterisk send a INVITE in-dialog<br>
to re-establish the audio (Dan Cropp)<br>
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----------------------------------------------------------------------<br>
<br>
Message: 1<br>
Date: Fri, 24 May 2019 17:02:56 +0000<br>
From: Dan Cropp <<a href="mailto:dan@amtelco.com" target="_blank">dan@amtelco.com</a>><br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a>><br>
Subject: Re: [asterisk-users] Is there a way to make asterisk send a<br>
INVITE in-dialog to re-establish the audio<br>
Message-ID:<br>
<<a href="mailto:dabd7263f5bb401f95b3375f70d8960e@AM-Mail2012B.amtelco.com" target="_blank">dabd7263f5bb401f95b3375f70d8960e@AM-Mail2012B.amtelco.com</a>><br>
Content-Type: text/plain; charset="utf-8"<br>
<br>
Thank you Joshua<br>
<br>
<br>
-----Original Message-----<br>
From: asterisk-users <<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>> On Behalf Of Joshua C. Colp<br>
Sent: Friday, May 24, 2019 9:53 AM<br>
To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>
Subject: Re: [asterisk-users] Is there a way to make asterisk send a INVITE in-dialog to re-establish the audio<br>
<br>
On Fri, May 24, 2019, at 9:47 AM, Dan Cropp wrote:<br>
> <br>
> We are working with an Avaya switch.<br>
> <br>
> <br>
> We send them a REFER. If the transfer is successful, everything is <br>
> great. If it fails (busy), they send an INVITE in-dialog with a media <br>
> attribute of inactive. After that, they send a 486 busy.<br>
> <br>
> The problem is Avaya basically put the call on hold so audio is not active.<br>
> <br>
> The Avaya rep is indicating we need to send in dialog invite to get <br>
> the call audio back? They are essentially saying they put the call on <br>
> hold because we told them to transfer and it’s our responsibility to <br>
> take the call off hold.<br>
> <br>
> <br>
> Is there a way to do this?<br>
<br>
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.<br>
<br>
[1] <a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_PJSIP_SEND_SESSION_REFRESH" rel="noreferrer" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_PJSIP_SEND_SESSION_REFRESH</a><br>
<br>
--<br>
Joshua C. Colp<br>
Digium - A Sangoma Company | Senior Software Developer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: <a href="http://www.digium.com" rel="noreferrer" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" rel="noreferrer" target="_blank">www.asterisk.org</a><br>
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</blockquote></div>