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<I><FONT SIZE="2">14:38:01.229941 IP 192.168.4.240.sip > 192.168.4.248.sip: SIP, length: 889</FONT></I><BR>
<I><FONT SIZE="2">14:38:01.230127 IP 192.168.4.248.sip > 192.168.4.240.sip: SIP, length: 515</FONT></I><BR>
<I><FONT SIZE="2">14:38:01.251558 IP 192.168.4.240.sip > 192.168.4.248.sip: SIP, length: 497</FONT></I><BR>
<I><FONT SIZE="2">14:38:01.271714 IP 192.168.4.240.sip > 192.168.4.248.sip: SIP, length: 1060</FONT></I><BR>
<I><FONT SIZE="2">14:38:01.271904 IP 192.168.4.248.sip > 192.168.4.240.sip: SIP, length: 433</FONT></I><BR>
<I><FONT SIZE="2">14:38:01.272133 IP 192.168.4.248.sip > 192.168.4.242.sip: SIP, length: 861</FONT></I><BR>
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is what I see... only SIP, no RTP/UDP...<BR>
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I guess you're right...<BR>
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Thank you, Tom.<BR>
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<BR>
On Sat, 2009-04-18 at 06:50 -0400, Tom Moore wrote:
<BLOCKQUOTE TYPE=CITE>
<FONT SIZE="2"><FONT COLOR="#0000ff">Asterisk still controls the signalling, but the audio path should be going through the phones directly.</FONT></FONT>
</BLOCKQUOTE>
<BLOCKQUOTE TYPE=CITE>
<FONT SIZE="2"><FONT COLOR="#0000ff">Fire up a tcpdump on the Asterisk server to varify this.</FONT></FONT><BR>
</BLOCKQUOTE>
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<BLOCKQUOTE TYPE=CITE>
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