[asterisk-users] Transfer to fax

Mike Diehl mdiehl at diehlnet.com
Tue Mar 13 16:28:36 CDT 2012


On Tuesday 13 March 2012 3:21:58 pm Danny Nicholas wrote:
> #1 you might need a progress() statement after answer

I'll try that.  Thank you.

> #2 what does sip show peer xxx look like on this peer?

I'm testing against my office phone, a Polycom 501:

  * Name       : 0004F211F1D0-2
  Realtime peer: Yes, cached
  Secret       : <Set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : customers
  Subscr.Cont. : <Not set>
  Language     : 
  Accountcode  : 1
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    : 
  Pickupgroup  : 
  Mailbox      : 7001 at context
  VM Extension : asterisk
  LastMsgsSent : 32767/65535
  Call limit   : 0
  Dynamic      : Yes
  Callerid     : "Mike Diehl" <5051234567>
  MaxCallBR    : 384 kbps
  Expire       : 172
  Insecure     : no
  Nat          : Always
  ACL          : Yes
  T.38 support : Yes
  T.38 EC mode : FEC
  T.38 MaxDtgrm: -1
  DirectMedia  : Yes
  PromiscRedir : No
  User=Phone   : No
  Video Support: Yes
  Text Support : No
  Ign SDP ver  : Yes
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 
  Addr->IP     : 173.10.242.192 Port 1811
  Defaddr->IP  : 0.0.0.0 Port 5060
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 0004F211F1D0-2
  SIP Options  : (none)
  Codecs       : 0x4 (ulaw)
  Codec Order  : (ulaw:20)
  Auto-Framing :  No 
  100 on REG   : Yes
  Status       : OK (88 ms)
  Useragent    : PolycomSoundPointIP-SPIP_501-UA/3.1.4.0070
  Reg. Contact : sip:0004F211F1D0-2 at 10.0.1.81
  Qualify Freq : 60000 ms
  Variables    :
                 line_id = 0004F211F1D0-2
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  Parkinglot   : 



> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mike Diehl
> Sent: Tuesday, March 13, 2012 4:18 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Transfer to fax
> 
> So I'm still trying to get this to work... (I'm top posting, but the
> details are below, if you want/need background info)
> 
> I'd like Asterisk to detect incoming faxes and redirect them elsewhere. 
> The details aren't important, as long as I get the detection working.
> 
> I've added this to my sip.conf file.  Probably overkill, but I'll tune it
> once it works:
> 
> [general]
> faxdetect=both
> 
> My sip registrations are all in a Mysql RT database, so I added this column
> to my table:
> 
> faxdetect char(3) default 'no'
> 
> I've set faxdetect to 'yes' for the devices that I expect to be receiving
> fax calls.
> 
> I did a sip reload from the console after adding and updating this column.
> 
> I've added a fax extension to the appropriate context in extensions.conf:
> exten => fax,1,noop(I hear a fax!)
> 
> Since most of my dialplan is in an AGI script, I've added this to the code
> that handles my test number:
> 
> $main::agi->answer();
> $main::agi->exec("ringing");
> $main::agi->exec("wait","5");
> 
> 
> So, now that all of this is in place, I call the extension from my fax
> machine... and I don't get any indication on the console that Asterisk
> heard a fax.  My extension simply rings and I answer it.
> 
> What am  missing?
> 
> TIA,
> Mike Diehl.
> 
> On Friday 24 February 2012 4:22:07 pm Kevin P. Fleming wrote:
> > On 02/24/2012 05:20 PM, Mike Diehl wrote:
> > > On Friday 24 February 2012 4:06:19 pm Kevin P. Fleming wrote:
> > >> On 02/24/2012 05:00 PM, Mike Diehl wrote:
> > >>> On Friday 24 February 2012 3:17:04 pm Mike Diehl wrote:
> > >>>> On Friday 24 February 2012 2:39:48 pm Kevin P. Fleming wrote:
> > >>>>> On 02/24/2012 03:32 PM, Mike Diehl wrote:
> > >>>>>> Hi all,
> > >>>>>> 
> > >>>>>> I've got a user that has one phone number an wants to be able
> > >>>>>> to us it for both voice and fax.
> > >>>>>> 
> > >>>>>> When a fax call comes in, he wants to do some incantation on
> > >>>>>> the keypad and have the call go to the fax machine.
> > >>>>>> 
> > >>>>>> As I see it, he has 3 options.
> > >>>>>> 
> > >>>>>> 1.  (blind?) Transfer it to the fax extension.
> > >>>>>> 
> > >>>>>> 2.  Use features.conf to create a key sequence, say *2, to
> > >>>>>> dial/transfer to a fax extension.
> > >>>>>> 
> > >>>>>> 3.  Use fax detect (SIP) to do it automatically.  However I'm
> > >>>>>> also using FFA, so I suspect these are mutually exclusive.
> > >>>>> 
> > >>>>> They are not. Enabling faxdetect should do exactly what you
> > >>>>> want; it will redirect the call to the 'fax' extension in the
> > >>>>> current context, and you can then Dial() your FAX machine (or
> > >>>>> send the call to ReceiveFAX).
> > >>>> 
> > >>>> Thank you.  Then, that's what I'll do.
> > >>> 
> > >>> On second though, I think my suggestion that FFA and fax detect
> > >>> were mutually exclusive stemmed from the idea that a call that was
> > >>> being originated/answered/handled by FFA would have it's call
> > >>> disconnected and redirected by fax detect.
> > >>> 
> > >>> If this is the case, it changes my dial plan logic, and I'm not
> > >>> sure I fully understand what changes I'll need to make.
> > >>> 
> > >>> For all I know, it might even simplify things by isolating all fax
> > >>> handling in one block.
> > >> 
> > >> Well, first you should not have faxdetect enabled on outbound
> > >> channels. That takes care of the 'originating' part.
> > >> 
> > >> If you have an inbound channel that you *know* you are sending to
> > >> ReceiveFAX, then you can just disable faxdetect on that channel
> > >> before doing so (this is why we made 'faxdetect' configurable from
> > >> the dialplan). Alternatively, you can just let calls that you
> > >> *know* are going to go to ReceiveFAX (dedicated FAX DIDs, for example)
> 
> just 'idle'
> 
> > >> in the dialplan listening to silence until faxdetect kicks in and
> > >> sends them to ReceiveFAX.
> > >> 
> > >> Note that the usage of FFA is not relevant here; whether you are
> > >> using Fax for Asterisk, the free version of it, or res_fax_spandsp,
> > >> the behavior and scenarios would be the same.
> > > 
> > > Very nice.
> > > 
> > > Sounds like I need to add a faxdetect column to my SIP real-time
> > > configuration. Once I've done a sip reload or pruned/loaded my user
> > > agents, I should be good to go!
> > > 
> > > Didn't know faxdetect was configurable in the dialplan...  Pointer
> > > to how to do it?
> > 
> > The CHANNEL() dialplan function with the 'faxdetect' option. Not sure
> > which releases have it; it might only be Asterisk 10.

-- 

Take care and have fun,
Mike Diehl.



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