<div dir="ltr"><br><div class="gmail_extra"><div class="gmail_quote">On Wed, Jul 3, 2013 at 1:31 PM, Carlos Chavez <span dir="ltr"><<a href="mailto:cursor@telecomabmex.com" target="_blank">cursor@telecomabmex.com</a>></span> wrote:<br>
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I have an Asterisk 11.4 SIP only system. We are using a SIP trunk<br>
for outside calls. We are having a problem with calls dropping after<br>
a transfer.<br>
<br>
Outside call awswered by phone 101<br>
101 transfers to 100 (attended transfer)<br>
call is dropped after a few seconds<br>
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I cannot really think of anything else to check in sip.conf.<br>
Incoming calls never drop if they are not transferred.<br>
<br></blockquote><div><br></div><div style>What does Asterisk say when the transfer occurs?</div><div><br></div><div style>You can also look at a trace of the SIP messages during the transfer using 'sip set debug on <peer>' (set it for both the transferer as well as the transfer destination). That should show why the requests are rejected and/or why a call is hungup.<br>
</div></div><div><br></div>-- <br><div dir="ltr"><div>Matthew Jordan<br></div><div>Digium, Inc. | Engineering Manager</div><div>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA</div><div>Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a></div>
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