Nikolay, I would think we would want both options...
Here in the US there is a need to be able to record incoming & outgoing without notification for purposes of investigations, but also for companies to do it as a matter of policy (which usually requires an announcement, but not always... if one party of a call knows it is being recorded that is enough) Mike On Mon, Oct 18, 2010 at 6:32 AM, Nikolay Kondratyev <[email protected]> wrote: > Mike, > this will record all incoming calls for a particular user. > There is no indication that the call is being recorded... But i'm not sure > .... Do you mean something like MWI? > I think, that voice prompt, notifying that a call is being recorded is > quite enough... > May be it's worth implementing such notification via FS "recorder" > dialplan, using playback application. > Rgds, > Nikolay. > > > > ------------------------------ > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Michael Picher > *Sent:* Friday, October 15, 2010 9:54 PM > > *To:* Discussion list for users of sipXecs software > *Subject:* Re: [sipx-users] trial call recording in sipx > > Sounds like a very useful addition Nikolay! > > Would this be set to record all calls for a particular user? Is there an > indication to the user that the call is being recorded (other than if you > were to capture SIP packets and really see what is going on)? > > Thanks, > Mike > > On Thu, Oct 14, 2010 at 7:52 AM, Nikolay Kondratyev <[email protected]> wrote: > >> Hi all, >> acording to my investigations in the tracker, call recording feature is >> delayed for indefinite time. >> Call recording function is highly anticipated. >> Meanwhile FS gives a toolkit to record voice. >> >> So i tried to do something myself. >> >> My main idea is to route a call, that is to be recorded, through a >> separate FS profile/dialplan. >> The idea is based on >> http://wiki.sipfoundry.org/display/xecsuserV4r2/Custom+FreeSWITCH+programming >> >> The main problem is: what is the creteria and how to route to that special >> FS profile and then route the call back to the subscriber? >> >> At the moment i found a partial solution (or a draft for a partial >> solution), i.e. how to record all _incoming_ calls for a specific >> subscriber. >> >> Here is a test configuration, that works (that is all calls to specific >> user are recorded in a wav file) for me: >> >> My sip domain is sip.nstel.ru, sipx hosthame is beaver.sip.nstel.ru >> Ok, i created separate sofia profile, listening on port 15085. >> For the 3800 user (i used this extension for testing, ) i set call >> forwarding with the following destination: [email protected]. >> sipxconfig does not allow to specify port in call forward destination, so >> i had to create dns srv records for _sip._udp.record.sip.nstel.ru, >> pointing to the same host, but port 15085, where recording FS profile is >> listening. >> I route the call back to >> [email protected];sipx-noroute=VoiceMail;sipx-userforward=false. Thus >> avoiding routing loop. >> Here is the FS dialplan for it: >> >> <extension name="record_call"> >> <condition field="destination_number" expression="^(3\d{3})$"> >> <action application="set" data="RECORD_TITLE=Recording >> ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> >> <action application="set" data="RECORD_COPYRIGHT=(c) 1980 Factory >> Records, Inc."/> >> <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/> >> <action application="set" data="RECORD_ARTIST=Ian Curtis"/> >> <action application="set" data="RECORD_COMMENT=Love will tear us >> apart"/> >> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d >> %H:%M)}"/> >> <action application="set" data="RECORD_STEREO=true"/> >> <action application="record_session" >> data="/var/sipxdata/mediaserver/data/recorder/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> >> <action application="set" data="ringback=${us-ring}"/> >> <action application="answer"/> >> <action application="sleep" data="300"/> >> <action application="bridge" data=" >> sofia/recorder/[email protected];sipx-noroute=VoiceMail;sipx-userforward=false"/<sofia/recorder/[email protected];sipx-noroute=VoiceMail;sipx-userforward=false%22/> >> > >> </condition> >> </extension> >> Please find attached xml trace of a recorded call. >> >> So... a kind of recording works and looks to be acceptable for at list >> further testing. >> >> I see two shortcomings in my method: >> >> 1. When the call is forwarded, it is not diverted, it is forked. That is >> sipxproxy sends invite to both 3800's phone and "recorder". >> Then when "FS recorder" answers a call, sipxproxy cancels "frame 31 >> invite" branch (see attached xml). This canceled branch is excessive. >> So the question: Is there a way to get rid of invite in frame 31? In other >> words, is it possible to divert a call instead of forking it? >> >> 2. Looking at frame 57, i see that spesial creadentials (~~id~media) are >> used by "FS recorder" to pass through "407 proxy authentication required. >> This user is configured in >> /etc/sipxpbx/freeswitch/conf/sip_profiles/sipX_profile.xml file. It is >> configured for a different profile, listening on another port. >> I don't understand why this user is used by "recorder" profile. >> I think that after some careful FS configuration this can be overcomed, >> though i don't think, this is the first problem in the queue. >> >> Next step: storing voice recordings in plain wav files will not be very >> convinient. It'll be difficult to find required file after a couple of >> months.... >> So next step could be: create separate postgres database, and store >> recording there... i guess this should be possible via mod_perl. >> And then create a php driven web page for accessing those records. >> >> Some questions to the community to summarize my efforts: >> 1. How do you find the idea? Do you think it can suite for a production >> system? >> 2. Can you foresee any problems? >> 3. Is it possible to divert a call instead of forking? >> 4. I did not managed to solve "route outgoing call from a specific user >> through a recorder" task. The problem is that, user should dial the same >> number, as usually. Any ideas? >> >> Thanks and regards, >> NIkolay. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > > > > -- > There are 10 kinds of people in this world, those who understand binary and > those who don't. > > [email protected] > blog: http://www.sipxecs.info > call: sip:[email protected] <sip%[email protected]> > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- There are 10 kinds of people in this world, those who understand binary and those who don't. [email protected] blog: http://www.sipxecs.info call: sip:[email protected] <sip%[email protected]>
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