<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:'times new roman', 'new york', times, serif;font-size:12pt"><div>Thanks a lot Kevin for the reply</div><div><br></div><div style="font-family:times new roman, new york, times, serif;font-size:12pt"><br><div style="font-family:arial, helvetica, sans-serif;font-size:13px"><font size="2" face="Tahoma"><hr size="1"><b><span style="font-weight: bold;">From:</span></b> Kevin P. Fleming <kpfleming@digium.com><br><b><span style="font-weight: bold;">To:</span></b> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br><b><span style="font-weight: bold;">Sent:</span></b> Thu, April 29, 2010 5:43:15 AM<br><b><span style="font-weight: bold;">Subject:</span></b> Re: [asterisk-users] No change in payload. (SDP)<br></font><br>
Aditya Kumar wrote:<br>> re-posting the question.<br>> -----------<br>> use case:<br>> when some one in my pbx calls 100.200, I have translations well defined,<br>> Media also (media via asterisk) --Works.<br>> when some one calls bob, or for any names I am adding Domain and call is<br>> been sent to the other party -- Works, no media...<br>> <br>> For the cases when it is talking to the external work,<br>> I want Astersik not to do anything with the SDP/payload.<br>> I want it to send as it is to the external proxy.<br>> <br>> How can I achieve this? so that the SDP/payload will not be modified for<br>> users talking to the external world.<br>> I want media for those external devices to come Directly to the users<br>> in my pbx. (with out going t asterisk)<br>> <br>> 2) also related question is can I have the xml payload in the originator<br>> and call is routed via PBX to the
Target.<br>> The xml payload also must be carried to the target.<br>> is it possible....<br>> <br>> This will really help me as I was held up with this :(<br><br>Neither of these are possible; Asterisk is a B2BUA, not a proxy, and as<br>such the outgoing INVITE is a *different* session from the incoming one.<br>That means that Asterisk has to be able to understand the SDP content<br>that arrives so it can forward media between the two sessions.<br><br>-- <br>Kevin P. Fleming<br>Digium, Inc. | Director of Software Technologies<br>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>skype: kpfleming | jabber: <a ymailto="mailto:kfleming@digium.com" href="mailto:kfleming@digium.com">kfleming@digium.com</a><br>Check us out at <a target="_blank" href="http://www.digium.com">www.digium.com</a> & <a target="_blank" href="http://www.asterisk.org">www.asterisk.org</a><br><br>--
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