<div dir="ltr"><div>Hi <font face="Tahoma">Prashanth / Bob,</font></div><div><font face="Tahoma"><br></font></div><div><font face="Tahoma">Avaya CM supports sip but this is mainly used for Aura VM, Experience portal and SM and more of Avaya only devices, direct conenctions to Asterisk , FS & 3CX work but I always experienced issues due to Avaya SIP Limitations and also officially this is not supported</font></div><div><font face="Tahoma">Best Route to do this is via session Manager as this is the supported method.</font></div><div><font face="Tahoma">First create a trunk link between Session manager and Freeswitch, then register extension directly to SM and make calls between from this extension and Freeswitch, once everything is working then create a trunk between CM & SM, or use existing and setup digit routing on SM and CM.</font></div><div><font face="Tahoma"><br></font></div><div><font face="Tahoma">if you have any issues setting up SM and its entities then let me know. this way will be best and far most easy one.</font></div><div><font face="Tahoma"><br></font></div><div><font face="Tahoma">thanks, kaleem</font></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Sep 8, 2015 at 9:54 AM, <span dir="ltr"><<a href="mailto:freeswitch-users-request@lists.freeswitch.org" target="_blank">freeswitch-users-request@lists.freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Send FreeSWITCH-users mailing list submissions to<br>
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<br>Today's Topics:<br>
<br>
1. Re: OneWay audio after hold->unhold (Bote Man)<br>
2. Re: Compiling under SmartOS (Stanislav Sinyagin)<br>
3. Re: OneWay audio after hold->unhold (Prashanth Devarajappa)<br>
<br><br>---------- Forwarded message ----------<br>From: Bote Man <<a href="mailto:bote_radio@botecomm.com">bote_radio@botecomm.com</a>><br>To: "'FreeSWITCH Users Help'" <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Date: Mon, 7 Sep 2015 18:03:17 -0400<br>Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold<br><div lang="EN-US" vlink="purple" link="blue"><div><p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Actually, yes I need all of the settings on the Avaya side. <u></u><u></u></span></p><p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></p><p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">I am not using encryption and I never could get the Avaya CS1000M to talk to the FreeSWITCH installation. I think the closest I could get was the signaling would allow the call to be answered, but I got no audio to pass either direction. <u></u><u></u></span></p><p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></p><p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">I think the Avaya puts information in unexpected elements of the SIP message and I had no more time to troubleshoot it because it was not my primary task.<u></u><u></u></span></p><p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></p><p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">There must be some small setting that I am missing and I would love to find the missing piece of the puzzle.<u></u><u></u></span></p><p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></p><p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Thanks!<u></u><u></u></span></p><p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></p><p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Bote<u></u><u></u></span></p><p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></p><p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></p><div style="border-width:medium medium medium 1.5pt;border-style:none none none solid;border-color:currentColor currentColor currentColor blue;padding:0in 0in 0in 4pt"><div><div style="border-width:1pt medium medium;border-style:solid none none;border-color:rgb(181,196,223) currentColor currentColor;padding:3pt 0in 0in"><p class="MsoNormal"><b><span style="font-family:"Tahoma","sans-serif";font-size:10pt">From:</span></b><span style="font-family:"Tahoma","sans-serif";font-size:10pt"> Prashanth Devarajappa<br><b>Sent:</b> Monday, 07 September, 2015 09:47<br><b>Subject:</b> Re: [Freeswitch-users] OneWay audio after hold->unhold<u></u><u></u></span></p></div></div><p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal"><b><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Avaya PBX Version Control<u></u><u></u></span></b></p><p class="MsoNormal" style="margin-left:1.25in"><span style="color:rgb(31,73,125);font-family:Wingdings;font-size:9pt">§</span><span style="color:rgb(31,73,125);font-size:7pt"> </span><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Media encryption over IP:Y</span><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u><u></u></span></p><p class="MsoNormal" style="margin-left:1.25in"><span style="color:rgb(31,73,125);font-family:Wingdings;font-size:9pt">§</span><span style="color:rgb(31,73,125);font-size:7pt"> </span><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Session Manager Element Manager 6.3<u></u><u></u></span></p><p class="MsoNormal" style="margin-left:1.25in"><span style="color:rgb(31,73,125);font-family:Wingdings;font-size:9pt">§</span><span style="color:rgb(31,73,125);font-size:7pt"> </span><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">System Manager Release 6.3.5<u></u><u></u></span></p><p class="MsoNormal" style="margin-left:1.25in"><span style="color:rgb(31,73,125);font-family:Wingdings;font-size:9pt">§</span><span style="color:rgb(31,73,125);font-size:7pt"> </span><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Communication System Management 6.3.10<u></u><u></u></span></p><p class="MsoNormal"><b><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></b></p><p class="MsoNormal"><b><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">In IP-Network-region :<u></u><u></u></span></b></p><p class="MsoNormal" style="margin-left:1.25in"><span style="color:rgb(31,73,125);font-family:Wingdings;font-size:9pt">§</span><span style="color:rgb(31,73,125);font-size:7pt"> </span><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Codec set matches group for encryption</span><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u><u></u></span></p><p><span style="color:rgb(31,73,125);font-family:Wingdings">§</span><span style="color:rgb(31,73,125);font-family:"Times New Roman","serif";font-size:7pt"> </span><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Intra-region IP-IP Direct Audio: yes<u></u><u></u></span></p><p><span style="color:rgb(31,73,125);font-family:Wingdings">§</span><span style="color:rgb(31,73,125);font-family:"Times New Roman","serif";font-size:7pt"> </span><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Inter-region IP-IP Direct Audio: yes<u></u><u></u></span></p><p class="MsoNormal"><b><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></b></p><p class="MsoNormal"><b><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">In CM, signaling-group of type</span></b><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">: <b>SIP</b><u></u><u></u></span></p><p class="MsoNormal" style="margin-left:1.25in"><span style="color:rgb(31,73,125);font-family:Wingdings;font-size:9pt">§</span><span style="color:rgb(31,73,125);font-size:7pt"> </span><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Transport Method: TLS<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></p><p class="MsoNormal"><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Let me know if you need further Avaya config details.<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></p><p class="MsoNormal"><b><span style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">FreeSwitch :<u></u><u></u></span></b></p><p class="MsoNormal"><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Its standard FS with custom (code/config) changes to enable <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></p><p><span lang="EN-GB" style="color:rgb(31,73,125);font-family:Symbol;font-size:11pt">·</span><span lang="EN-GB" style="color:rgb(31,73,125);font-size:7pt"> </span><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Support for RFC 5939 : Capability Negotiation<u></u><u></u></span></p><p><span lang="EN-GB" style="color:rgb(31,73,125);font-family:Symbol;font-size:11pt">·</span><span lang="EN-GB" style="color:rgb(31,73,125);font-size:7pt"> </span><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Support for UNENCRYPTED_SRTCP<u></u><u></u></span></p><p><span lang="EN-GB" style="color:rgb(31,73,125);font-family:Symbol;font-size:11pt">·</span><span lang="EN-GB" style="color:rgb(31,73,125);font-size:7pt"> </span><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Support IP Shuffling with in ACK…… to be done </span><span lang="EN-GB" style="color:rgb(31,73,125);font-family:Wingdings;font-size:11pt">J</span><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></p><p class="MsoNormal"><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></p><p class="MsoNormal"><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt">Prashanth<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></p><p class="MsoNormal"><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></p><p class="MsoNormal"><span lang="EN-GB" style="color:rgb(31,73,125);font-family:"Calibri","sans-serif";font-size:11pt"><u></u> <u></u></span></p><p class="MsoNormal"><b><span style="font-family:"Calibri","sans-serif";font-size:11pt">From:</span></b><span style="font-family:"Calibri","sans-serif";font-size:11pt"> Bote Man [<a href="mailto:bote_radio@botecomm.com" target="_blank">mailto:bote_radio@botecomm.com</a>] <br><b>Sent:</b> 05 September 2015 04:50<br><b>To:</b> FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br><b>Subject:</b> Re: [Freeswitch-users] OneWay audio after hold->unhold<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB"><u></u> <u></u></span></p><div><div><div><p class="MsoNormal" style="margin-bottom:12pt"><span lang="EN-GB">We would be VERY appreciative if you could provide the configurations on the Avaya and FreeSWITCH that allow them to talk to each other. In my experience Avaya does not play well with others, and I have tried and given up.<u></u><u></u></span></p></div><p class="MsoNormal" style="margin-bottom:12pt"><span lang="EN-GB">Thank you in advance!<u></u><u></u></span></p></div><p class="MsoNormal" style="margin-bottom:12pt"><span lang="EN-GB">Bote<u></u><u></u></span></p></div><div><p class="MsoNormal"><span lang="EN-GB"><u></u> <u></u></span></p><div><p class="MsoNormal"><span lang="EN-GB">On Fri, Sep 4, 2015 at 12:09 PM, Prashanth Devarajappa <<a href="mailto:Prashanth.Devarajappa@enghouse.com" target="_blank">Prashanth.Devarajappa@enghouse.com</a>> wrote:<u></u><u></u></span></p><blockquote style="border-width:medium medium medium 1pt;border-style:none none none solid;border-color:currentColor currentColor currentColor rgb(204,204,204);margin:5pt 0in 5pt 4.8pt;padding:0in 0in 0in 6pt"><div><div><p class="MsoNormal"><span lang="EN-GB">Hello,<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB"> <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB">I am connecting to Avaya switch using SIP trunk interface from FS with TLS & SRTP connection. This works well for inbound and outbound calls. I have an issue with media, when I try hold -> unhold operation from FS for an outbound call.<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB"> <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB">When FS tries to unhold the call by sending Re-INVITE, avaya responds with 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, to make media flow directly b/w endpoints). FS responds with 200 OK + SDP as expected. Now avaya sends ACK + SDP where SDP has the updated connection info(IP and port of endpoint) where media should be sent. Along with this it also changes the crypto master key as per RFC due to change in connection info(c=).<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB"> <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB">On seeing this ACK, FS update the connection info(which is good) and change its own crypto master key in response. however this new master key is not sent to far end, meaning FS can decode the media it receives but other end can't decode the media FS is sending to it resulting in one way audio.<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB"> <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB">Has anyone come across this issue ? <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB"> <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB">Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets new master key ?<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB"> <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB"> <u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB">Regards<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB" style="color:rgb(136,136,136)">Prashanth<u></u><u></u></span></p><p class="MsoNormal"><span lang="EN-GB" style="color:rgb(136,136,136)"> <u></u><u></u></span></p></div></div><p class="MsoNormal"><span lang="EN-GB"><br>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a><br><a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br><br>Official FreeSWITCH Sites<br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br><a href="http://confluence.freeswitch.org" target="_blank">http://confluence.freeswitch.org</a><br><a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br><br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><u></u><u></u></span></p></blockquote></div><p class="MsoNormal"><span lang="EN-GB"><u></u> <u></u></span></p></div></div></div></div><br><br>---------- Forwarded message ----------<br>From: Stanislav Sinyagin <<a href="mailto:ssinyagin@gmail.com">ssinyagin@gmail.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Date: Tue, 8 Sep 2015 08:55:37 +0200<br>Subject: Re: [Freeswitch-users] Compiling under SmartOS<br>See the update at <a href="https://freeswitch.org/jira/browse/FS-7967" target="_blank" rel="noreferrer">https://freeswitch.org/jira/browse/FS-7967</a><br>
<br>
I fixed the compilation problems, and now there's a runtime issue.<br>
<br>
On Mon, Aug 17, 2015 at 10:40 AM, Stanislav Sinyagin<br>
<<a href="mailto:ssinyagin@gmail.com">ssinyagin@gmail.com</a>> wrote:<br>
> I see there are some people on the list, working with SmartOS.<br>
><br>
> The current master fails to compile:<br>
> <a href="https://freeswitch.org/jira/browse/FS-7967" target="_blank" rel="noreferrer">https://freeswitch.org/jira/browse/FS-7967</a><br>
><br>
> Your input will be appreciated.<br>
><br>
> I just started looking around and getting the feeling what SmartOS is.<br>
> I worked with Solaris quite a lot, but that was almost 10 years ago.<br>
<br>
<br>
<br><br>---------- Forwarded message ----------<br>From: Prashanth Devarajappa <<a href="mailto:Prashanth.Devarajappa@enghouse.com">Prashanth.Devarajappa@enghouse.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Date: Tue, 8 Sep 2015 08:53:42 +0000<br>Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold<br>
<div lang="EN-GB" vlink="purple" link="blue">
<div>
<p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri",sans-serif;font-size:11pt">Hi Bob,<u></u><u></u></span></p>
<p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri",sans-serif;font-size:11pt"><u></u> <u></u></span></p>
<p class="MsoNormal" style="-ms-text-autospace:;"><span style="color:rgb(31,73,125);font-family:"Calibri",sans-serif;font-size:11pt">I am told that our Avaya is “</span><span style="color:black;font-family:"Segoe UI",sans-serif;font-size:10pt">avaya</span><span style="color:rgb(31,73,125);font-family:"Calibri",sans-serif;font-size:11pt">
</span><span style="color:black;font-family:"Segoe UI",sans-serif;font-size:10pt">communication manager is the 8300”</span><span style="font-size:11pt"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri",sans-serif;font-size:11pt"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri",sans-serif;font-size:11pt">So far with our testing, we have not seen any issues related to transfer. (May be we have not done extensive transfer tests)<u></u><u></u></span></p>
<p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri",sans-serif;font-size:11pt"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri",sans-serif;font-size:11pt">Prashanth<u></u><u></u></span></p>
<p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri",sans-serif;font-size:11pt"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="color:rgb(31,73,125);font-family:"Calibri",sans-serif;font-size:11pt"><u></u> <u></u></span></p>
<p class="MsoNormal"><b><span lang="EN-US" style="font-family:"Calibri",sans-serif;font-size:11pt">From:</span></b><span lang="EN-US" style="font-family:"Calibri",sans-serif;font-size:11pt"> Bob Hartwig [mailto:<a href="mailto:bobjectsfreeswitch@gmail.com" target="_blank">bobjectsfreeswitch@gmail.com</a>]
<br>
<b>Sent:</b> 07 September 2015 17:42<br>
<b>To:</b> FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
<b>Subject:</b> Re: [Freeswitch-users] OneWay audio after hold->unhold<u></u><u></u></span></p>
<p class="MsoNormal"><u></u> <u></u></p>
<div>
<p class="MsoNormal">Is this an Avaya AS5300 by any chance? Those have some serious interop issues when it comes to transfers. I had to struggle mightily to get a client's product (not Freeswitch) to interoperate with it. I'm really interested to hear if
this gets resolved for Freeswitch.<u></u><u></u></p>
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<p class="MsoNormal"><u></u> <u></u></p>
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<p class="MsoNormal"><u></u> <u></u></p>
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<p class="MsoNormal"><u></u> <u></u></p>
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<p class="MsoNormal">On Fri, Sep 4, 2015 at 11:09 AM, Prashanth Devarajappa <<a href="mailto:Prashanth.Devarajappa@enghouse.com" target="_blank">Prashanth.Devarajappa@enghouse.com</a>> wrote:<u></u><u></u></p>
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<p class="MsoNormal">Hello,<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal">I am connecting to Avaya switch using SIP trunk interface from FS with TLS & SRTP connection. This works well for inbound and outbound calls. I have an issue with media, when I
try hold -> unhold operation from FS for an outbound call.<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal">When FS tries to unhold the call by sending Re-INVITE, avaya responds with 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, to make media flow directly b/w endpoints).
FS responds with 200 OK + SDP as expected. Now avaya sends ACK + SDP where SDP has the updated connection info(IP and port of endpoint) where media should be sent. Along with this it also changes the crypto master key as per RFC due to change in connection
info(c=).<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal">On seeing this ACK, FS update the connection info(which is good) and change its own crypto master key in response. however this new master key is not sent to far end, meaning FS
can decode the media it receives but other end can't decode the media FS is sending to it resulting in one way audio.<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal">Has anyone come across this issue ?
<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal">Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets new master key ?<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal">Regards<u></u><u></u></p>
<p class="MsoNormal"><span style="color:rgb(136,136,136)">Prashanth<u></u><u></u></span></p>
<p class="MsoNormal"><span style="color:rgb(136,136,136)"> <u></u><u></u></span></p>
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<p class="MsoNormal"><br>
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