Hi Jason, I repetedly tested the scenario: Skype->skypopen->FS->SIP->FS and this mens, a skype call incoming to FS and then bridged via SIP to another FS you can test the same on your machine if you make the destination for skypopen interface pointing to the extension 5000 of the default dialplan (the standard IVR) and then press 1. Can you please test the 5000 ext press 1, and confirm that it works (or not) for you? Can you then post (better in a Jira) the exact way to replicate your problem? (dialplan, actions, configs, whatever) Thanks for reporting, -giovanni On Tue, Aug 24, 2010 at 1:34 AM, Jason Jeffords <jason@cloudtree.net> wrote:
Hi Giovanni, Our case 3 test was actually being bridged to a SIP endpoint (not in a conference, although this probably should not matter). We tested two type 3 cases: 1) Skype to FreeSWITCH Skype bridged to an extension of a registered SIP phone 2) Skype to FreeSWITCH Skype bridged to an outbound call through a PSTN gateway In both cases we are transiting FreeSWITCH, not terminating on it (could there be a race condition when doing signaling coordination to remote SIP endpoints, not FreeSWITCH itself?). When we terminate Skype calls on FreeSWITCH this works (case 2). It also works for outbound Skype calls (case1 - SIP to FreeSWITCH, FreeSWITCH Skype to Skype). Also, we are running the very latest (well, as of this morning) git version, so that may introduce one more variable (if you are running an older version). Thanks for your help, Jason
On Mon, Aug 23, 2010 at 6:36 PM, Giovanni Maruzzelli <gmaruzz@celliax.org> wrote:
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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