Ahh! ok. I understand now. You're going to have a PSTN->SIP gateway -> this dictation recorder sipXtapi UA.
This dictation UA will need to handle (from what I've been hearing from you), many incoming calls - initially fine with 16, but final product will need to handle 50-100 or more simultaneous calls. You want a way to record said calls from the dictation UA itself, and you want to interpret DTMF tones. This all sounds perfectly do-able by sipXtapi. It'll require a bunch of work at the application level to perform the recording (and/or see Jaro's record patch), you'll need to do something based on the DTMFs received - this also is easy to do with sipXtapi, afaik. You'll set up your UA so it has 1 AOR, but multiple internal lines, and when a call comes in, you'll transfer it to one of the internal lines for handling. When new calls come in, you'll do the same, up to the maximum (currently 16, but can be changed in code). Once the maximum is hit, the sipXtapi call manager will send (I believe) a 486 (busy indicator) back to the remote user (doctor). On 5/10/07, Daniel Sigurgeirsson <[EMAIL PROTECTED]> wrote:
>> How exactly is your dictation software working with regards to calls? I'm not sure I completely understand your question, but the flow of the "conversation" is as follows: - doctor calls a certain number my software is monitoring - the software gets a notification about the call, finds out who is calling, and plays the appropriate introductory message. - the doctor enters the patient SSN throught the phone keypad, as well as other information (the type of the dictation report etc.) - when all the necessary numeric information which can be entered through the keypad are done, the doctor starts recording his report and my software records his voice. - during the recording the doctor can use the keypad to control the recording, just like if he were using a regular dictaphone (pause, play, rewind, ffwd etc.) - when the recording is done all these information are transferred to another computer. I believe that the only basic requirements I have are: - be able to play sound files to the user (the doctor) - be able to receive the voice from the user (and record it) - be able to intercept the DTMF tones I had a solution which used standard TAPI, but the Alcatel phone station in question didn't support everything we needed, hence I'm investigating using SIP/RTP instead. Regards, Daníel ------------------------------ Date: Wed, 9 May 2007 14:56:44 -0400 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [sipxtapi-dev] Multiple simultaneous calls to the same number CC: [email protected] The default maximum number of calls is 16, but there are people out there using sipXtapi with upwards of 100 calls or more (there have been reports of someone receiving several hundred calls). If you wish to receive more than 16 calls, you will have to change: #define MAX_MANAGED_FLOW_GRAPHS 16 to some new value. I also believe that the mixer currently can mix only 10 calls, but this also can be easily changed. How exactly is your dictation software working with regards to calls? I don't entirely follow what sort of requirements you need from sipXtapi or other SIP stack. On 5/9/07, *Daniel Sigurgeirsson* < [EMAIL PROTECTED]> wrote: My application will not be dialling, it will only receive phone calls. Ideally it should allow many users calling at the same time, presumably from different phone numbers. The application in question is a dictation solution for doctors, where doctors call this particular number to dictate a report which is then transferred to a secretary. It would be very limiting if only one doctor could dictate at the same time... Regards, Daníel ------------------------------ > To: [email protected] > Date: Wed, 9 May 2007 18:54:49 +0200 > From: [EMAIL PROTECTED] > Subject: [sipxtapi-dev] Multiple simultaneous calls to the same number > > >Hi, > > > >I just recently started looking into using SIP for a dictation application I'm developing. It should allow users to call a single number, >and I would like to be able to allow more than one user to call at the same time. The upper limit should of course be as high as >possible!, but a few (around 16 perhaps) would suffice to begin with. What are my options here? Should I create X many lines, each in >a separate thread, and then forward calls to the main number to the next available line? Or do I need to run many instances of the >program simultaneously? Is this perhaps not possible using SIP? Any input would be most welcome, since this is still quite new to >me. > > Do you want to dial from the same number or from many different numbers? If you need from just one, then 1 line will suffice. You can make multiple calls on the same line. Line URLs passed to sipxLineAdd have to be unique. > > Jaro > _______________________________________________ > sipxtapi-dev mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/ ------------------------------ Get news, entertainment and everything you care about at Live.com<http://live.com/>. Check it out! <http://www.live.com/getstarted.aspx+> _______________________________________________ sipxtapi-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/ -- Keith Kyzivat SIPez LLC. SIP VoIP, IM and Presence Consulting http://www.SIPez.com <http://www.sipez.com/> tel: +1 (617) 273-4000 ------------------------------ Get news, entertainment and everything you care about at Live.com. Check it out! <http://www.live.com/getstarted.aspx+> _______________________________________________ sipxtapi-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipxtapi-dev/
-- Keith Kyzivat SIPez LLC. SIP VoIP, IM and Presence Consulting http://www.SIPez.com tel: +1 (617) 273-4000
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