it sounds as if the FS SBC is bridging the call and creating separate forks, and allowing them to be picked up individually instead of making this a single stream.
it has a lot of security implications in doing so, IMO. You might want to revisit how your FS b2bua is configured and how the bridge is established. I feel reasonably sure there would be a way to keep this from happening, but I don't know this would be a good example of how to establish a b2bua connection, given the multiple connections to a single stream. Have you asked about this on the fs-users list? Since this is not a direct configuration by sipx, it's certainly not a bug. The question remains is whether it is a feature request! On Tue, Oct 18, 2011 at 9:09 AM, Fulvio Scapin <[email protected]>wrote: > Hello Kumaran. > > I had forgotten to mention my SipXECS version in my previous mail. I > fear it's still a 4.2. Haven't found the time to upgrade yet. > Your scenario appears mostly correct, although I didn't quite get the > first three points you enumerated. > > Just to add a few more details, I use FreeSWITCH as SBC in a B2BUA > configuration, connected to a few ITSP SIP providers. > So what actually happens is that I receive a SIP invite through my > ITSP account, which FreeSWITCH «bridges» with my SipXECS default AA. > When the AA answers the caller digits an internal extension number and > the call is transferred to that extension, which would be the 200 of > your example. > If nobody answers at 200 and user 201 dials *78200 as you imagined, > the call is instantiated solely between the external caller and > extension 201, as one might expect. > > However, if the user at 200 picks up the call after the user at 201 > has dialed *78200, they BOTH answer the call, in somethin akin to a > three-way conference. > > I do realize it's not intended to behave that way, but it might become > a useful feature rather than a bug for quite a few people, hopefully. > > Regards, > Fulvio > > 2011/10/18 Kumaran T <[email protected]>: > > Hi Fulvio, > > Which Build you seeing this issue? > > I tried in latest 4.5.2 build by following scenario and its working > > fine.Please let me know my scenario was right? > > 1.DID no assigned to AA > > 2.Call DID number from PSTN number > > 3.AA will answer and dial 200 from pstn number > > 4.The call will transferred to 200 and 200 will start ringing > > 5.From 201 user dialed *78200 > > 6. Call is retrieved and call will disconnected from 200 > > 7.Call will established between pstn user and 201.. > > > > Regards, > > Kumaran T > > > > On 10/18/2011 1:22 PM, Fulvio Scapin wrote: > >> Hello again. > >> I've been finally able to reproduce in a repeatable way the phenomenon > >> I described a few weeks ago. > >> > >> Situation: > >> > >> Call from outside to the SipXECS-based IVR, dialing the internal > >> extension directly through the IVR, let's say extension 31. > >> > >> Extension 31 (polycom soundpoint 550) rings. > >> > >> The operator at extension 30 (also polycom soundpoint 550) dials *7831 > >> to pick up the call. > >> After he has dialed *7831 the user at extension 31 picks up the call > >> and begins talking. > >> A second or two later the user at extension 30 picks up the call as > well. > >> > >> End result: > >> > >> A three-way call in which each of three parties involved can speak > >> with and hear the other two. > >> > >> > >> Am I the only one finding this peculiar? > >> > >> Bye, > >> Fulvio Scapin > >> _______________________________________________ > >> sipx-users mailing list > >> [email protected] > >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > _______________________________________________ > > sipx-users mailing list > > [email protected] > > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net <http://support.myitdepartment.net>Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services!
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