Thanks for the response.. I did change it in the [general] settings.My setup is something like I have a remote conference (not meetme) which will send reinvite to redirect the RTP flow to a different server to load balance.There are three clients who join in the conference and i can listen to two other clients speak from the third client but when i record the conversation my recording of one of the clients ends before the stipulated hangup time. I am guessing this is because one of the clients doesn't understand what to do with a reinvite.. Any suggestions.In the SIP.conf i have changed the directmedia option to no and also enabled the ignoresdpversion option.<br>
<br><div class="gmail_quote">On Tue, Jul 3, 2012 at 10:01 PM, SamyGo <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I don't think you can set SIP properties in some variables anywhere in asterisk dialplan or call file. What you can do is change the directmedia options of the SIP or any other channel you're using. i.e if your call file has<div>
<br></div><div>CHANNEL=SIP/12345@latestgateway</div><div> <br>Then change the properties of the [latestgateway] in sip.conf. Also if you're using an IP address directly</div><div><br></div><div>CHANNEL=SIP/<a href="mailto:12345@10.10.4.4" target="_blank">12345@10.10.4.4</a></div>
<div><br></div><div>Then you can change the directmedia directive in sip.conf [general] settings.</div><div><br></div><div>Hope it helped.</div><div><br></div><div>BR</div><div>Sammy Go.</div><div><br></div><div><div class="gmail_quote">
<div class="im">
On Wed, Jul 4, 2012 at 2:08 AM, sathiish kumar <span dir="ltr"><<a href="mailto:sathiish.kumar@gmail.com" target="_blank">sathiish.kumar@gmail.com</a>></span> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="im">
I am using call files to make calls to a remote machine but can't seem to quite understand the directmedia options that are set by default in Asterisk.Is there any way i can specify the directmedia options using call files?<br clear="all">
<div><br></div><br>
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