On Wed, Jul 3, 2013 at 1:31 PM, Carlos Chavez <[email protected]>wrote:
> -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > I have an Asterisk 11.4 SIP only system. We are using a SIP trunk > for outside calls. We are having a problem with calls dropping after > a transfer. > > Outside call awswered by phone 101 > 101 transfers to 100 (attended transfer) > call is dropped after a few seconds > > I cannot really think of anything else to check in sip.conf. > Incoming calls never drop if they are not transferred. > > What does Asterisk say when the transfer occurs? 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. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org
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