[asterisk-users] Calls drop after transfer

[Matthew Jordan]

On Wed, Jul 3, 2013 at 1:31 PM, Carlos Chavez <cursor at telecomabmex.com>wrote:

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>         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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