On Wed, Jan 21, 2009 at 1:49 AM, Ognjen Seslija <[email protected]> wrote: > When call comes in from Openzap, tone_detect app does pre_answer of a call > cause it's need media to start detecting tones in the first place. This > behaviour is something that I see on calls inside my telco when calling from > analogue lines. I don't think this is a big of deal because ringback > provided by FS will make caller understand that the call is still in > progress. One can make its own ringback to sound exactly the same as > telco's. > > I don't think that we'll ever make POTS behave like digital protocols do.
So true! -MC > > Regards, > Ognjen > > On Wed, Jan 21, 2009 at 10:36 AM, Scott Ellis <[email protected]> > wrote: >> >> I had a similar problem, you can use >> <action application="set" data="ringback=${au-ring}"/> (I added an "au" >> ring definition to my vars.xml file) >> >> To get what you want. >> >> I also had a problem that you get two rings, then an answer then to the >> system generated ring tone, which was confusing some of our (not to bright) >> callers. >> >> As we don't use callerID I turned that flag off in the openzap.conf.xml >> file - I thought that this would do what I wanted (answer the instant the >> call is detected), but the change in the config file does not make it all >> the way down to the point where it takes action. At this point I hacked the >> code to get what I wanted. I have to create a JIRA entry with the details >> yet. >> >> As far as I understand, this is the right place for OpenZap, as it is a >> product of the FS project. >> >> Scott >> >> Tomás wrote: >> >> Scott, I imagined that it could be an OpenZap problem, but I didn't find >> an OpenZap mailing list, so I sent the email to FS list. Do you know where >> can I find more information about OpenZap hardware support and developement >> status (I have special interest in Loop Start)?? >> >> Anthony and Ognjen, I've tried tone detection and thanks to that FS is >> detecting hung up, but I faced the problem that tone detector answer the >> call... >> >> That's my dialplan: >> >> <extension name="extension_name"> >> <condition field="destination_number" expression="^919999999$"> >> <action application="tone_detect" data="busy 425,0 r +100 hangup >> 16 4"/> >> <action application="bridge" >> data="sofia/internal/1003%${server-domain-name}, >> sofia/internal/1004%${server-domain-name}"/> >> </condition> >> </extension> >> >> When I receive a call from PSTN, tone detection answer the call (the >> caller hears only one first tone and then hears "nothing" until I pick up >> the call on softphone). >> >> So, I think that tone detection solution does not resolve my problem... Is >> there any other possibility to detect hang up without answering the call >> (using Loop Start signaling) or have we to wait until OpenZap is completely >> developed? >> >> Thanks in advance. >> >> On Tue, Jan 20, 2009 at 10:43 PM, Ognjen Seslija <[email protected]> >> wrote: >>> >>> Ok, as discussed with Tony on IRC channel I followed his directions which >>> lead to a successfull outcome (like it always does I might add :). >>> >>> One has to use tone_detect app in FreeSWITCH dialplan in order to check >>> for busy tones coming from the PSTN side and if matched fire a hangup >>> application. This is the snippet of my test dp that does the trick (from >>> extension Local_extensions in default.xml): >>> >>> <anti-action application="tone_detect" data="busy 425,0 r +100 hangup 16 >>> 4"/> >>> <anti-action application="bridge" >>> data="user/${dialed_extensi...@${domain_name}"/> >>> This means that FS will listen to freq of 425 Hz and wait for 4 positive >>> detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425 >>> Hz is the freq telco here uses; for other countries I suggest getting the >>> ITU world tones pdf file and check there): >>> >>> 2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 >>> tone_detect_callback() TONE busy HIT 1/4 >>> 2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 >>> tone_detect_callback() TONE busy HIT 2/4 >>> 2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 >>> tone_detect_callback() TONE busy HIT 3/4 >>> 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 >>> tone_detect_callback() TONE busy HIT 4/4 >>> 2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 >>> tone_detect_callback() TONE busy DETECTED >>> 2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup >>> OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING] >>> >>> Regards, >>> Ognjen >>> >>> On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija <[email protected]> >>> wrote: >>>> >>>> I tried similar setup with my analog card (X100P) and I'm having same >>>> issue. Call is not hungup on the oz side once the caller ends. My telco >>>> doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck >>>> to detecting busy tone from the telco side. I'll try to modify tones.conf >>>> accordingly. >>>> >>>> Regards, >>>> Ognjen >>>> (sekil) >>>> On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale >>>> <[email protected]> wrote: >>>>> >>>>> This is a common issue with analog phones even traditional answering >>>>> machines suffer from it. >>>>> I'm sure you must have had an answering machine at some point that has >>>>> dial tone as the message it receives. >>>>> >>>>> Unless FreeSWITCH has some hint that the call has hungup it will not >>>>> stop trying to complete the call. >>>>> >>>>> If the other side is sending a busy tone to indicate hangup it's >>>>> possible to use the tone_detect app to pick >>>>> up on the tones and abort the call. >>>>> >>>>> Another thing you could do if you have unlimited inbound is explicitly >>>>> answer the call in the dialplan before >>>>> you call your sip phones this will give you a more profound hangup >>>>> detection but it will make every call count >>>>> even when nobody answers. >>>>> >>>>> >>>>> >>>>> On Tue, Jan 20, 2009 at 10:46 AM, Tomás <[email protected]> >>>>> wrote: >>>>>> >>>>>> Hi all, >>>>>> >>>>>> I'm configuring my home PBX using FreeSwitch. I'm using a X101P card >>>>>> configured as FXO (conected to analog PSTN line) and I have several IP >>>>>> phones and softphones conected to FreeSwitch. >>>>>> >>>>>> I can call from an IP phone to other IP phone (the same with the >>>>>> softphones) and also from an IP phone (or softphone) to an external >>>>>> number >>>>>> thought PSTN. >>>>>> >>>>>> When I call from an external analog phone to FreeSwitch, I bridge the >>>>>> call to all internal IP phones and softphones and they ring, but the >>>>>> problem >>>>>> is that when I hang up the call in the external phone, all internal >>>>>> phones >>>>>> (IP phones and softphones) keeps ringing... >>>>>> >>>>>> I'm pretty sure the problem is that FreeSwitch don't receive the hang >>>>>> up, because I cann't see anything on the log. >>>>>> >>>>>> I've also created my own tones.conf for my country (Spain) but I'm not >>>>>> sure if it's ok (but I have the same problem with hang up) >>>>>> >>>>>> I've googled the list, and I've found several people with a similar >>>>>> problem but no solution... >>>>>> >>>>>> That's my pastebin with the most importants printouts and config >>>>>> files: >>>>>> http://pastebin.freeswitch.org/6822 >>>>>> >>>>>> Thank you very much in advance. >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> [email protected] >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> AIM: anthm >>>>> MSN:[email protected] >>>>> GTALK/JABBER/PAYPAL:[email protected] >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:[email protected] >>>>> iax:[email protected]/888 >>>>> googletalk:[email protected] >>>>> pstn:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> [email protected] >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> [email protected] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> ________________________________ >> _______________________________________________ >> Freeswitch-users mailing list >> [email protected] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> [email protected] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
