January 2011 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Processing Live Audio during a call between 2 extensions
by Michel Habib 16 Jan '11

16 Jan '11

Dear Freeswitch Developers, i am relatively new to freeswitch and i am seeking some advice. I am creating an External Application that does some [live] processing on the Audio of the freeswitch call before sending it to the second Call Leg, and vice versa. Can you please direct me to the best way to do that and how to access/capture the live audio channel in both legs while it is streaming, and how to resubmit it again to the other leg? I read the book about freeswitch, but still couldnt figure the best approach to do that, any hint is appreciated. Best Regards, Michel Habib.

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[Freeswitch-dev] FreeSWITCH 1.0.7
by Brian West 14 Jan '11

14 Jan '11

http://latest.freeswitch.org/ Enjoy! /b

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[Freeswitch-dev] Fwd: Peculiar behavior in FreeSWITCH when media streams for SRTP and RTP are offered in the same SDP
by Goutham BG 14 Jan '11

14 Jan '11

Sending the below query to FreeSWITCH-dev list as well. Any hints will be very helpful. Thanks Goutham B G ---------- Forwarded message ---------- From: Goutham BG <bggoutham(a)gmail.com> Date: Wed, Jan 12, 2011 at 4:56 PM Subject: Peculiar behavior in FreeSWITCH when media streams for SRTP and RTP are offered in the same SDP To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Hi, I am observing a peculiar behavior in FreeSWITCH-1.0.7 with SRTP. I have the following entry in my dialplan XML file: <extension name="IVR"> <condition field="destination_number" expression="^IVR$"> <action application="set" data="sip_secure_media=true"/> <action application="bridge" data="loopback/app=socket: 47.152.232.156:8084 async full"/> </condition> </extension> A Polycom SoundPoint IP 550 configured in "SRTP best effort" mode dials into this extension and is connected to the IVR. In this mode, the phone offers two media streams in the SDP of INVITE; 1st one for SRTP and the 2nd one for RTP. But the problem is that the media is established in SRTP in one way and RTP in the other way. The phone offers the following SDP in the INVITE message: v=0 o=- 1167766638 1167766638 IN IP4 47.152.232.149 s=Polycom IP Phone c=IN IP4 47.152.232.149 t=0 0 a=sendrecv m=audio 5040 RTP/SAVP 9 0 8 18 127 a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:svWhF9Uh0VonfhtmuvRLM4B9S6+ZEicc3hDd3dAQ a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 m=audio 5040 RTP/AVP 9 0 8 18 127 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 As we can see, there are two "m=" lines in the SDP of the offer; the 1st one for SRTP and the other one for RTP. FreeSWITCH-1.0.7 answers the call by sending 200OK with the following SDP: v=0 o=FreeSWITCH 1294817064 1294817065 IN IP4 47.152.232.156 s=FreeSWITCH c=IN IP4 47.152.232.156 t=0 0 m=audio 11552 RTP/SAVP 9 127 a=rtpmap:9 G722/8000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=silenceSupp:off - - - - a=ptime:20 a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:z5/m/v2U0negVFjfkCsXR/sNAo9fhBv+dspBIbtx m=audio 0 RTP/AVP 19 As we can see above, FreeSWITCH accepts the SRTP stream and rejects the RTP stream (by sending port as 0) in the SDP. The Polycom phone sends the media in SRTP as expected. But, FreeSWITCH sends the media in RTP to the phone even though it accepted SRTP in the answer (200OK). Please let me know if this is a bug or am I missing something here? The debug log of the call coming into FreeSWITCH is pasted here http://pastebin.freeswitch.org/14998 . If required I can also send the wireshark traces of this scenario captured in the system where FreeSITCH is running. Thanks Goutham B G

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[Freeswitch-dev] session:setHanhupHook is launched with session:tranfer
by Arturo Monroy 13 Jan '11

13 Jan '11

Hi, session:setHangupHook is execute when is executed "transfer" api, however the "session" is not hanging and session still live? Should be considered a bug? Shouldn't be lauched when session is terminate by CS_DESTROY EVENT? I'm using Lua, and using fifo too, after a several X seconds the current session is transfer-out to lua application, acording script's tree, the current session could come-back or not again to FIFO, however the hangupHook API is executed each one the session:transfer is execute. -- Saludos Arturo Monroy (55) 5371-1100 Ext 189 ASR Arturo Lopez Auronix de Mexico www.auronix.mx

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[Freeswitch-dev] FreeSWITCH Conference Call Today
by Michael Collins 12 Jan '11

12 Jan '11

Hello all, We have a light agenda today: http://wiki.freeswitch.org/wiki/FS_weekly_2011_01_12 Bring your questions and ideas and be ready to share! -MC

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[Freeswitch-dev] FreeSWITCH on github
by Seven Du 12 Jan '11

12 Jan '11

Is this official? https://github.com/FreeSWITCH/FreeSWITCH Is it there to provider an alternative repo to download code or as a backup or even accept pull requests? Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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[Freeswitch-dev] FreeSWITCH Conference Call Today
by Michael Collins 05 Jan '11

05 Jan '11

Hey all! Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_01_05 We have a few things to discuss. Also, if DRK's schedule permits he will be dropping by to talk more about his billing stuff! Hope to talk to you soon. -Michael

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[Freeswitch-dev] PlayAndGetDigits backward compatibility
by babak yakhchali 03 Jan '11

03 Jan '11

Hi I was trying to build my old scripts with latest revision in mod_managed and I found out PlayAndGetDigits takes one more argument: public string PlayAndGetDigits(int min_digits, int max_digits, int max_tries, int timeout, string terminators, string audio_files, string bad_input_audio_files, string digits_regex, string var_name, *int digit_timeout*); Isn't it possible to add an overload for backward compatibility? thanks

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Re: [Freeswitch-dev] FreeBSD build problem
by Kim Culhan 02 Jan '11

02 Jan '11

On Sat, January 1, 2011 11:39 pm, Oleg Khovayko wrote: > Hi, > > During last build from GIT version on FreeBSD v7.2, > I see following error message: > > {{{ > > In file included from libs/spandsp/src/plc.c:47: > libs/spandsp/src/spandsp/saturated.h: In function 'fsaturate': > libs/spandsp/src/spandsp/saturated.h:129: warning: implicit declaration > of function 'lrintl' > libs/spandsp/src/spandsp/saturated.h:129: warning: incompatible implicit > declaration of built-in function 'lrintl' > gmake[1]: *** [libfreeswitch_la-plc.lo] Error 1 > gmake: *** [all] Error 2 > > }}} > > I checked source, > File: libs/spandsp/src/spandsp/saturated.h > > Line #129 contains: > return (int16_t) lrint((float)damp); > > > I assume, somewhere in the headers is redefinition lrint -> lrintl, > and FreeBSD can not found this function. > > > I substituted that problem line by: > > 131 return (int16_t) lrintf((float)damp); > > And now everything builds OK. FWIW git of this date builds without this problem on FreeBSD 8.2-RC1. -kim

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[Freeswitch-dev] FreeBSD build problem
by Oleg Khovayko 02 Jan '11

02 Jan '11

Hi, During last build from GIT version on FreeBSD v7.2, I see following error message: {{{ In file included from libs/spandsp/src/plc.c:47: libs/spandsp/src/spandsp/saturated.h: In function 'fsaturate': libs/spandsp/src/spandsp/saturated.h:129: warning: implicit declaration of function 'lrintl' libs/spandsp/src/spandsp/saturated.h:129: warning: incompatible implicit declaration of built-in function 'lrintl' gmake[1]: *** [libfreeswitch_la-plc.lo] Error 1 gmake: *** [all] Error 2 }}} I checked source, File: libs/spandsp/src/spandsp/saturated.h Line #129 contains: return (int16_t) lrint((float)damp); I assume, somewhere in the headers is redefinition lrint -> lrintl, and FreeBSD can not found this function. I substituted that problem line by: 131 return (int16_t) lrintf((float)damp); And now everything builds OK. You can apply this fix in the your GIT source tree. Thanks, Oleg

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