Hi Jason, with 3) - the problematic one - you mean an incoming call from a remote skype to the fs-skype, that is then bridged to an outbound sip call? I'm asking this because this is the most tested case, eg: in the gateway for the FS-conference, where the incoming skype calls are then bridged (through dialplan, like pressing "1" in the standard IVR) with the sip conference. And it works with no problems... Let me know more, please. -giovanni On Mon, Aug 23, 2010 at 8:28 PM, Jason Jeffords <jason@cloudtree.net> wrote:
Hi Giovanni, We got some test cycles today and have found the following: 1) SIP to Skype calls work as desired - A-leg is SIP, B-leg is Skype 2) Skype to Internal Extensions Work as desired - A-leg is Skype, B-leg is FreeSwitch (internal extension) 3) Skype to SIP does not work - A-leg is Skype, B-leg is SIP In the last case: 1) We hear Skype ringing until the B-leg (SIP) is answered - this is correct 2) We then get dead air and the SIP B-leg hangs up on its own (i.e. the user answering does not hang up, but the connection hangs up) 3) The Skype client connection stays up with dead air until it is manually hung up We are running the latest code as of today. Any insight you could provide will be most appreciated. Thanks of all your work, Jason On Fri, Aug 13, 2010 at 5:31 PM, Giovanni Maruzzelli <gmaruzz@celliax.org> wrote:
Jason,
please test with the latest git version.
Many messy changes to an already messy code, so maybe some bug or side effect (ok, bug) has been introduced.
commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090 Author: Giovanni Maruzzelli <gmaruzz@gmail.com> Date: Fri Aug 13 16:19:20 2010 -0500
skypopen: now answer a call only when directed to do it (before was trying to answer any incoming call). Lot of changes to a messy part, so maybe some problem will come out...
Signed-off-by: Giovanni Maruzzelli <gmaruzz@gmail.com>
-giovanni -- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
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-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618