sipx call records conferences is configured to do so. polycom 650's have a usb port that can be used to record a call from the phone if they buy the productivity license from polycom.
because internal calls have the media going peer-to-peer you will either need to span a port that sipx sits on and use a separate system to record calls or you will need to create a service/role (and probably a dedicated server) that will be able to receive multicast streams. These streams will need to be defined somewhere (phone?) in order to send their audio to their peer/gateway and to the recording server/role. I think this would be easier implemented using a port span and a separate server. the disadvantage is that if sipx is anchoring your media with sipxbridge, you will get those audio streams but not your internal (non remote) users. You have to wonder how the phones will behave if they multicast every call and what that will do to bandwidth and quality. In either instance, a separate machine should be used (IMO). On Fri, Aug 26, 2011 at 6:44 AM, Fulvio Scapin <[email protected]>wrote: > All the UAs where Polycom SoundPoint IP 550. I don't remember what the > firmware version was at the time. > Honestly, at the time we didn't want to confuse the client calling us, > so we didn't exactly explore the situation. > However I think the call was full-duplex (and B didn't pick up the call). > > Btw, I know it's difficult to implement, especially if one uses an > external B2BUA instead of, or in parallel to sipXbridge, but are there > news or future projects about call recording? > As you might imagine, in a business/marketing environment, it could be > a very useful feature. > > Bye and thanks, > Fulvio Scapin > > 2011/8/13 Joegen Baclor <[email protected]>: > > This is definitely a race condition and will be very hard to reproduce. > If > > the calling UA got confused (or actually supports adhoc conferencing > > multiple early media channels) due to a fork then you will get exactly > what > > you have experienced. The trick is both 200 Ok must be close enough to > each > > other so as not to generate a CANCEL to one of the transactions. > > > > A couple of questions to determine if it's a case of a UA confused by > fork > > or a cool UA feature supporting multi channel forks: > > > > 1. is the audio heard by C and A full duplex? Meaning (C and A can talk > to > > each other) > > 2. Can both C and A be heard by B? > > > > if the answer to both Q's is yes, then you got a very cool UA. Send in > the > > brand. > > > > On 08/12/2011 09:01 PM, Fulvio Scapin wrote: > >> > >> Hello to everyone. > >> Just wishing to report a strange accident occured to me some time ago > >> (so forgive me were the details a bit vague), mostly out of curiosity > >> about possible similar experiences. > >> A small orientation about my setup: > >> FreeSWITCH as a B2B UA connected to serveral ITSPs, upstream from a > >> SipXecs proxy, redirecting incoming calls to an auto-attendant > >> embedded in SipXecs. > >> > >> The strange event happened as follows: > >> > >> * incoming calls directed to a hunt group of SipXecs after exiting the > >> default auto-attendant > >> * phone A not in the hunt group, phones B and C in the hunt group > >> * phones B and C ring > >> * phone A tries to pick up the call with *78 and extension B > >> * simultaneously phone C picks up the call > >> > >> Result: > >> the two sides of the call are heard by both C and A, as if, for > >> instance, one was monitoring C's call using phone A . > >> > >> Since call monitoring isn't a built-in feature of SipXecs (and my boss > >> said 'I want that' as soon as we understood what had taken place), I > >> was wondering whether mine was a freak accident or something more > >> promising. > >> > >> Regards, > >> 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/ > -- ====================== 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 faxservices!
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