April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

session.steamFile misses DTMF event for first few seconds
by Frank ＠ Impact 28 Apr '10

28 Apr '10

I recently upgraded from FS 12790M to svn 17188. When I did, I noticed that session.streamFile behaved differently and I started having problems with my IVR app. With the upgraded FS, I have a problem with streamFile no firing on the DTMF and calling the callback function for the first few seconds of the wav file playback. It behaves as though it does not hear the DTMFs. If I wait for 2 seconds or so of the wav file and then DTMF, streamFile catches the DTMF and all is well. If I key as soon as I hear the wav file start, streamFile just keeps playing the wav and does not call the callback function. When I revert back to the previous version of FS, streamFile always fires the callback right away no matter how quickly I press the first DTMF as the wav file starts to stream out. The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz The snippet of js code I am using is as follows. if(session.ready()) { session.answer(); session.sleep(750); while(session.ready()) { session.sleep(500); session.flushDigits(); // clear out input buffers if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi ts_cb,""))===false) { pin=session.getDigits(pinmax,pinterm,pinwait); } else { pin+=session.getDigits(pinmax-1,pinterm,pinwait); } // more code here.. } Do I need to change the way I use streamFile in the later release?

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Replace "Freeswitch" in SDP
by Ken Fulmer 28 Apr '10

28 Apr '10

This is a trivial request but here goes: We are using the User Agent parameter to replace the name "Freeswitch" with our company name in the SIP header. How can we do the same in the SDP? Thanks, Ken Fulmer

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Stupid question ;-)
by patrick 28 Apr '10

28 Apr '10

I have some var: <action application="set" data="input_fail_counter=0"/> And in some moment I want to increase it by 1. I try this way: <action application="set" data="input_fail_counter=${input_fail_counter}+1"/> And have result input_fail_counter=0+1 i.e. just string merge... How to handle it right way in dialplan? -- С уважением, Равальдини Патрик Стефанович Инженер по тестированию ООО «Центр речевых технологий» Тел.: (812) 325-8848, доб. 6736 Факс: (812) 327-9297 E-mail: patrick(a)speechpro.com http://www.speechpro.ru

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need registration help
by Travis Stevens 28 Apr '10

28 Apr '10

I am trying to register a sip account from one freeswitch (PBX) to an ITSP running freeswitch. The freeswitch at the ITSP tries to route the call but the contact info has prepended gw+ to the sip account registration. Is there a way to stop the gw+ from being prepended?

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mod_say_ru says time +1 hour of current time
by Boris Kovalenko 28 Apr '10

28 Apr '10

Hello! After looking to MODAPP-421 Michael Jerris wrote to me: "set timezone channel var" and closed the issue. So I changed my extension to: <extension name="SayTime"> <condition field="destination_number" expression="^9000$"> <action application="answer"/> <action application="set" data="timezone=Asia/Yekaterinburg"/> <action application="set" data="ct=${strftime_tz(Asia/Yekaterinburg %s)}"/> <action application="log" data="INFO TZ: ${timezone}"/> <action application="log" data="INFO CT: ${ct}"/> <action application="say" data="ru current_date_time pronounced ${ct}"/> <action application="sleep" data="1000"/> <action application="hangup"/> </condition> </extension> But pronounced time is still +1 hour (but ${ct} shows right time). What's wrong with my extension? Have I misunderstood Michael and need to set channel var in another place? -- Respect, Boris

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SIP Headers (e.g., Retry-After)
by Dan Le 28 Apr '10

28 Apr '10

Is there a way to get SIP headers like the Retry-After field in FreeSWITCH? Essentially we want to parse the Retry-After field and react based on the value for when we get 503s because of congestion or what not. Thanks, Dan

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E3 Card on Freeswitch ?
by Anita Hall 27 Apr '10

27 Apr '10

Hi Please check out this product http://www.sangoma.com/products/hardware_products/data_networking/a301.html Does it work on Freeswitch or Asterisk ? Do Telcos provide an E3 connection ? One of our customers had an inquiry for terminating 6000 calls simultaneously. I want to do some homework before taking it further with him. If I use E1 lines, I will need 6000 / 30 = 200 E1 lines, which does not look feasible ? Thanks for any input you may provide. regards, Anita Hall, Simmortel Voice.

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FreeSWITCH Community Conference Call April 28
by Michael Collins 27 Apr '10

27 Apr '10

Hello all! I've posted the agenda for tomorrow's call: http://wiki.freeswitch.org/wiki/FS_weekly_2010_04_28 Please get in there and add your items. It's been a bit hectic for me of late so I appreciate any and all contributors to the agenda. If you have a topic to discuss or a question to ask then definitely put it on the agenda so that we can be ready for it. Don't forget, we've pushed the meeting back one hour, so it starts at 1PM Eastern, 10AM Pacific (1700 UTC) See you all tomorrow! -Michael

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Re: [Freeswitch-users] mod_skinny - Building with VS 2008 Express
by Dave Stevenson 27 Apr '10

27 Apr '10

Mathieu, yes, it has made it into the SVN distribution now. I don't know why it was missing before, but it's there now regards Dave ----- Original Message ----- From: "Dave Stevenson" <stevendt(a)primrosebank.net> To: "Mathieu Parent" <math.parent(a)gmail.com> Sent: Tuesday, April 27, 2010 9:36 PM Subject: Re: [Freeswitch-users] mod_skinny - Building with VS 2008 Express > Hi Mathieu > > thanks for the reply - yes, I can see it in the Git link that you > included, but I'm currently using SVN. I don't know how far SVN is behind > Git at the moment, but the skinny_api files were not downloaded when I > last did an SVN update - I'll try again now and see if they have made it > to the SVN distribution yet. I will be able to download them from the link > in the meantime though > > regards > Dave > > > ----- Original Message ----- > From: "Mathieu Parent" <math.parent(a)gmail.com> > To: "Dave Stevenson" <stevendt(a)primrosebank.net> > Cc: <freeswitch-users(a)lists.freeswitch.org> > Sent: Tuesday, April 27, 2010 9:15 PM > Subject: Re: [Freeswitch-users] mod_skinny - Building with VS 2008 Express > > >> On Fri, Apr 23, 2010 at 10:23 PM, Dave Stevenson >> <stevendt(a)primrosebank.net> wrote: >>> Hi, >>> >>> I'm still struggling to get the VS2008 Express Build working, but there >>> seems to be another problem with mod_skinny >> I can't help much on VS 2k8 part but, >> >>> Mathieu, >>> >>> mod_skinny.c has a #include looking for skinny_api.h, the file does not >>> seem >>> to be there ? >> >> It is here. see >> http://fisheye.freeswitch.org/browse/freeswitch-git/src/mod/endpoints/mod_s… >> >> >> Mathieu >> >

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Route calls based on the Nb they are divert by
by Andreas Dröscher 27 Apr '10

27 Apr '10

Hi everyone I was wondering if it is possible route calls based on the numbers they were divert by, perhaps using "sip_redirected_by". Let's assume I own two phone numbers of my old fixed telephone lines and one VoIP Account/Number. If I redirect the two fixnet numbers to the same VoIP number, is it possible to "demultiplex" the incoming calls based on the number they were redirected by? I am not sure about the available meta data provided by SIP, however I am certain that my ISDN phone is sometimes showing "xxxx is calling, redirected by yyyyy" Best Wishes Andreas

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