<span style>Hi Jared & Kevin,</span><br style><br style><span style>Thanks for taking the time to answer my questions. I wonder if I could just be reading the tcpdump incorrectly? I'm still seeing rtp streams (and Jared, I have modified the dial string to remove the L)..</span><br style>
<br style><span style>Here's a screenshot of what I'm seeing in wireshark. I really appreciate the suggestions.</span><div style><br></div><div style>Screenshot: <a href="http://dl.dropbox.com/u/4156401/Screenshot%20from%202012-05-23%2007%3A39%3A51.png">http://dl.dropbox.com/u/4156401/Screenshot%20from%202012-05-23%2007%3A39%3A51.png</a> </div>
<div style><br></div><div style>pcap: <a href="http://dl.dropbox.com/u/4156401/trace3000.pcap">http://dl.dropbox.com/u/4156401/trace3000.pcap</a></div><div style><br></div><div style>Thanks</div><div style>David</div><br>
<div class="gmail_quote">On Wed, May 23, 2012 at 7:41 AM, David Wessell <span dir="ltr"><<a href="mailto:david@ringfree.biz" target="_blank">david@ringfree.biz</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi Jared & Kevin,<br><br>Thanks for taking the time to answer my questions. I wonder if I could just be reading the tcpdump incorrectly? I'm still seeing rtp streams (and Jared, I have modified the dial string to remove the L)..<br>
<br>Here's a screenshot of what I'm seeing in wireshark. I really appreciate the suggestions.<div><br></div><div>Thanks</div><div>David<div><div class="h5"><br><br><br><br>On Mon, May 21, 2012 at 6:08 PM, Jared Geiger <<a href="mailto:jared@compuwizz.net" target="_blank">jared@compuwizz.net</a>> wrote:<br>
> A2billing usually stays in the media path due to the dialstring<br>> parameters that it uses to cut a call off when the balance would reach<br>> $0. To get Asterisk to step out of the media path, I had to change<br>
> dialcommand_param_sipiax_friend and dialcommand_param to |60|S(14400)<br>> which lets all calls go to 14400 seconds. The default uses the L<br>> parameter. You need to use the S parameter instead. However the S<br>
> parameter doesn't like very large numbers in Asterisk 1.4 so I've just<br>> hard set mine.<br>><br>> ~Jared<br>><br>> On Mon, May 21, 2012 at 5:18 PM, Kevin P. Fleming <<a href="mailto:kpfleming@digium.com" target="_blank">kpfleming@digium.com</a>> wrote:<br>
>> On 05/21/2012 03:45 PM, David Wessell wrote:<br>>>><br>>>> More specific on sip.conf<br>>>><br>>>> In sip.conf I have a trunk specified for the SIP provider, and a trunk<br>>>> specified for the PBX itself.<br>
>>><br>>>> Do I need to specify directmedia=yes on both sides?<br>>><br>>><br>>> Yes, it has to be set on both peers involved in the bridged call.<br>>><br>>><br>>> --<br>
>> Kevin P. Fleming<br>>> Digium, Inc. | Director of Software Technologies<br>>> Jabber: <a href="mailto:kfleming@digium.com" target="_blank">kfleming@digium.com</a> | SIP: <a href="mailto:kpfleming@digium.com" target="_blank">kpfleming@digium.com</a> | Skype: kpfleming<br>
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>>> Check us out at <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
>><br>>> --<br>
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><br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>><br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
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