[Freeswitch-users] howto originate fs call from webapp (python)
hi all,
i come from asterisk an i am new to freeswitch. after my with days with
freeswitch i am very excited!
but trying to migrate our deployment i have three challenges. one of them is:
i need to call freeswitch from a webapp (e.g. python) and pass number1 and
number2. i then need freeswitch to call number1. as soon as it is picked up say
a short confirmaton text, call number2 and bridge the two.
my first approach was to call via xml_rpc like described in the wiki but when i
call like
server.freeswitch.api("originate","sofia/gategay/gateway1/{number1}
&bridge(sofia/gateway/gateway2/{number2})")
but in this case both numbers are called in parallel and the first number to
pick up gets a ringback tone until the other number picks up. how can i get the
sequence described above?
thanks for your help
alex
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Re: [Freeswitch-users] howto originate fs call from webapp (python)
hi,
oops, i must have been very tired when i wrote my first mail to the list...
thanks for your replies. {ignore_early_media=true} really worked for me.
i try very hard to "unlearn" asterisk.
with asterisk i did not do much more with the python script, but i would like
the pthon script to interact more with freeswitch like:
- call number1
- say a welcome message with cepstral voice
- call number2
- bridge
other scenario:
enter telephone number in webapp
python script have fs to call number
say "please enter the pin code from the website"
validate dtmf code
pass back to webapp: correct or not correct
unfortunately just from reading the wiki i don't know how to do it in my python
script.
can you share your experience?
thanks
alex
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[Freeswitch-users] problem speaking with cepstral voice in xml ivr menu
hi all, here is my second problem trying to migrate from * to fs: i can speak with cepstral voices from my dialplan, but when i implement an ivr menu with cepstral voices like this: i get the following errors: [ERR] mod_native_file.c:68 native_file_file_open() Error opening /usr/local/freeswitch/sounds/en/us/callie/say:text to speak.GSM can you point me in the right direction? thanks alex --- freeswitch 1.0.3 build 12166 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu
brian, thanks for your help. i really apreciate the active support. upgrading to the current trunk solved the problem. i can hear the cepstal voices in the ivr menus now. (with the current trunk i have all sorts of other compile problems (mod_fax, python) but i will work this out following the build instructions again). i only wonder because these worked with my last version of last week. so far i am a happy camper with freeswitch. this is a different snack bracket than asterisk... kind regards alex - Ursprüngliche Mail - Alex, If you want to update to svn trunk the tts-engine and tts-voice are now valid options on the menu. They were not before (But the wiki said they were). So to cut confusion I made them work... if you do not wish to upgrade you'll need to set the tts_engine and tts_voice variables before you call the IVR application and it will work with the code you already have. I highly recommend a "make curret" ;) Committed revision 12278. /b On Feb 25, 2009, at 4:35 AM, Alexander de Greiff wrote: > hi all, > > here is my second problem trying to migrate from * to fs: > > i can speak with cepstral voices from my dialplan, but when i > implement an ivr menu with cepstral voices like this: > > greet-long="say:text to speak" >greet-short="say:main menu" >invalid-sound="say:invalid entry" >exit-sound="say:goodbye" >timeout ="1" >max-failures="3" >tts-engine="cepstral" >tts-voice="allison" >phrase_lang="en"> > > > > > > i get the following errors: > > [ERR] mod_native_file.c:68 native_file_file_open() Error opening / > usr/local/freeswitch/sounds/en/us/callie/say:text to speak.GSM > > > can you point me in the right direction? > > thanks > alex > > > --- > freeswitch 1.0.3 build 12166 > ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu
michael, i googled for "asterisk alternative" and voila... the trigger was that every now and then i renew servers in the infrastructure and the one with asterisk was overdue. i wasn't really unhappy with asterisk, but these things bothered me (maybe i am not up to date): - dialplan gets messy - no conferences without hardware (rented remote server!) - ivr with cepstral voices: sometimes get hickups so far i like the fs approach very much. stable sip channels, no hickups with voices. kind regards alex - Ursprüngliche Mail - Von: "Michael Collins" An: freeswitch-users@lists.freeswitch.org Gesendet: Mittwoch, 25. Februar 2009 19:31:24 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu > so far i am a happy camper with freeswitch. this is a different snack bracket > than asterisk... If you don't mind telling us, where did you hear about FS and what made you decide to try it? Are you unhappy with Asterisk or are you simply looking for something a bit different? Just curious. Thanks, MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org 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 Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] dialplan condition regex question
hi all, i am dialing the number 123456789 (example) reaching fs via inbound sip gateway and hitting following dialplan: ... ... via info i can see that the variable my_dialed_extension is populated ok with 789 but somehow the second condition is not met. when i change that to match (.*) the actions gets executed and the my_dialed_extension inside is correct. any suggestions? kind regards alex ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] switch voices in ivr menus
another question: - in the ivr application i have cepstral voice matthias read the main menu ok. - i select submenu1 and cepstral voice katrin reads the submenu1 correctly. - i go back to the main menu and the voice is not switched back to the specified voice matthias. each voice is explicitly specified in each menu. any suggestions? kind regards alex ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] switch voices in ivr menus
brian, demo3 is the main menu. all submenus change voices correctly. when i go to the main menu via menu-sub (7) then the voice is changes correctly. only when i menu-top (9) to main menu the voice is not changed. how do i produce the debug log? in the cli? this i a remote terminal. f8 is not an option. here is the part of my ivr.conf.xml: ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
