<div class="gmail_quote"><br><br><div class="gmail_quote">On Sat, Jun 18, 2011 at 6:40 AM, Larry Moore <span dir="ltr"><<a href="mailto:lmoore@starwon.com.au" target="_blank">lmoore@starwon.com.au</a>></span> wrote:<br>
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<div>On 18/06/2011 5:36 AM, Matteo Campana wrote:<br>
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Inviato da iPhone<br>
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Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling<<a href="mailto:EWieling@nyigc.com" target="_blank">EWieling@nyigc.com</a>> ha scritto:<br>
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We experience the same thing. The solution we use is to not change codecs in the middle of a call. I assumed it was an issue with our upstream.<br>
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Hi Eric,<br>
this behavior is an asterisk bug or asterisk can never change the codec "on the fly"?<br>
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Thanks,<br>
Matteo<br>
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The problem reported occurs after a fax tone is detected, one might expect T.38 or G711 to be used to handle the fax, not G729!<br>
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There is no configuration file information for each of the nodes/peers, no debug of each peer involved nor a trace of the packets hence no one will have ideas!<br><font color="#888888">
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Larry.</font></blockquote><div><br></div><div><br></div><div>Hi Larry,</div><div>I have the SIP debug taken from asterisk.</div><div>In this debug: 1.2.3.4 ---> IP SIP PROXY </div><div> 5.6.7.8 ---> IP UAC (Linksys SPA 962)</div>
<div> 9.10.11.12 ---> IP ASTERISK to connect to the provider</div><div> 13.14.15.16 --> IP PROVIDER</div><div> 17.18.19.20 --> IP ASTERISK </div>
<div><br></div><div><br></div><div>The SIP debug is available at this link: <a href="http://pastebin.com/9DrFDmeC">http://pastebin.com/9DrFDmeC</a></div><meta http-equiv="content-type" content="text/html; charset=utf-8"><div>
<br></div><div><br></div><div>Thanks in advance,</div><div>Matteo</div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br>
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