From khovayko at gmail.com Sun Jan 2 07:39:44 2011 From: khovayko at gmail.com (Oleg Khovayko) Date: Sat, 01 Jan 2011 23:39:44 -0500 Subject: [Freeswitch-dev] FreeBSD build problem Message-ID: <4D200190.4080209@gmail.com> 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 From babak.freeswitch at gmail.com Mon Jan 3 15:03:27 2011 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Mon, 3 Jan 2011 15:33:27 +0330 Subject: [Freeswitch-dev] PlayAndGetDigits backward compatibility Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110103/245423eb/attachment.html From anthony.minessale at gmail.com Mon Jan 3 18:20:18 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 3 Jan 2011 09:20:18 -0600 Subject: [Freeswitch-dev] PlayAndGetDigits backward compatibility In-Reply-To: References: Message-ID: SWITCH_DECLARE(char *) playAndGetDigits(int min_digits, int max_digits, int max_tries, int timeout, char *terminators, char *audio_files, char *bad_input_audio_files, char *digits_regex, const char *var_name = NULL, int digit_timeout = 0); This is how it's declared in the c++ swig should pass this into its translation and fill in 0 when it's not specified. On Mon, Jan 3, 2011 at 6:03 AM, babak yakhchali wrote: > 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 > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From babak.freeswitch at gmail.com Mon Jan 3 18:50:38 2011 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Mon, 3 Jan 2011 19:20:38 +0330 Subject: [Freeswitch-dev] PlayAndGetDigits backward compatibility In-Reply-To: References: Message-ID: thanx for the answer but there is only one prototype in FreeSWITCH.Managed.dll which needs the last parameter and because of this when someone links to dll, on building, an error comes up which says there is no overload which .... this is the only signature from FreeSWITCH.Managed.dll metadata: 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); -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110103/3207aa66/attachment-0001.html From msc at freeswitch.org Wed Jan 5 20:50:58 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 5 Jan 2011 09:50:58 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110105/54307dcd/attachment.html From w8hdkim at gmail.com Sun Jan 2 17:07:46 2011 From: w8hdkim at gmail.com (Kim Culhan) Date: Sun, 2 Jan 2011 09:07:46 -0500 Subject: [Freeswitch-dev] FreeBSD build problem Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110102/fb040608/attachment.html From dujinfang at gmail.com Tue Jan 11 17:09:08 2011 From: dujinfang at gmail.com (Seven Du) Date: Tue, 11 Jan 2011 22:09:08 +0800 Subject: [Freeswitch-dev] FreeSWITCH on github Message-ID: 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 From brian at freeswitch.org Tue Jan 11 17:32:25 2011 From: brian at freeswitch.org (Brian West) Date: Tue, 11 Jan 2011 08:32:25 -0600 Subject: [Freeswitch-dev] FreeSWITCH on github In-Reply-To: References: Message-ID: <8DE7DC60-D618-4128-802E-41D311FFCE6F@freeswitch.org> I don't think we put that there... Mike? /b On Jan 11, 2011, at 8:09 AM, Seven Du wrote: > 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. > > -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110111/2e2d95c6/attachment.html From william.suffill at gmail.com Tue Jan 11 19:10:48 2011 From: william.suffill at gmail.com (William Suffill) Date: Tue, 11 Jan 2011 11:10:48 -0500 Subject: [Freeswitch-dev] FreeSWITCH on github In-Reply-To: <8DE7DC60-D618-4128-802E-41D311FFCE6F@freeswitch.org> References: <8DE7DC60-D618-4128-802E-41D311FFCE6F@freeswitch.org> Message-ID: https://github.com/freeswitch-mirrors/freeswitch is also a mirror of the FS tree. No idea who/how it got there tho. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110111/05b32322/attachment.html From msc at freeswitch.org Tue Jan 11 21:28:02 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 11 Jan 2011 18:28:02 +0000 Subject: [Freeswitch-dev] FreeSWITCH on github In-Reply-To: <8DE7DC60-D618-4128-802E-41D311FFCE6F@freeswitch.org> References: <8DE7DC60-D618-4128-802E-41D311FFCE6F@freeswitch.org> Message-ID: Andrew (Vagabond) and I put it there as an experiment. Every 15 min or so it updates, so the github is pretty much in lockstep with our repo. It's there as a backup in case we have issues with our servers. Some people just love their github so we gave them a way to get their FS fix... -MC On Tue, Jan 11, 2011 at 2:32 PM, Brian West wrote: > I don't think we put that there... Mike? > > /b > > On Jan 11, 2011, at 8:09 AM, Seven Du wrote: > > 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. > > -- > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110111/58f81a55/attachment.html From mike at jerris.com Tue Jan 11 22:16:18 2011 From: mike at jerris.com (Michael Jerris) Date: Tue, 11 Jan 2011 14:16:18 -0500 Subject: [Freeswitch-dev] FreeSWITCH on github In-Reply-To: References: <8DE7DC60-D618-4128-802E-41D311FFCE6F@freeswitch.org> Message-ID: <1DE3ACBB-0AE0-4E79-89D4-C17F010489BA@jerris.com> The official repo is: http://wiki.freeswitch.org/wiki/Installation_Guide#Git git://git.freeswitch.org/freeswitch.git Mike On Jan 11, 2011, at 1:28 PM, Michael Collins wrote: > Andrew (Vagabond) and I put it there as an experiment. Every 15 min or so it updates, so the github is pretty much in lockstep with our repo. It's there as a backup in case we have issues with our servers. Some people just love their github so we gave them a way to get their FS fix... > > -MC > > On Tue, Jan 11, 2011 at 2:32 PM, Brian West wrote: > I don't think we put that there... Mike? > > /b > > On Jan 11, 2011, at 8:09 AM, Seven Du wrote: > >> 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? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110111/93333848/attachment.html From andrew at hijacked.us Wed Jan 12 18:33:05 2011 From: andrew at hijacked.us (Andrew Thompson) Date: Wed, 12 Jan 2011 10:33:05 -0500 Subject: [Freeswitch-dev] FreeSWITCH on github In-Reply-To: References: <8DE7DC60-D618-4128-802E-41D311FFCE6F@freeswitch.org> Message-ID: <20110112153305.GF25773@hijacked.us> On Tue, Jan 11, 2011 at 06:28:02PM +0000, Michael Collins wrote: > Andrew (Vagabond) and I put it there as an experiment. Every 15 min or so it > updates, so the github is pretty much in lockstep with our repo. It's there > as a backup in case we have issues with our servers. Some people just love > their github so we gave them a way to get their FS fix... My personal reasons where I was sick of FishEye being down every time I went to use it, and its nice to follow FreeSWITCH along with the other projects I keep an eye on in github. Andrew From andrew at hijacked.us Wed Jan 12 18:49:25 2011 From: andrew at hijacked.us (Andrew Thompson) Date: Wed, 12 Jan 2011 10:49:25 -0500 Subject: [Freeswitch-dev] FreeSWITCH on github In-Reply-To: <20110112153305.GF25773@hijacked.us> References: <8DE7DC60-D618-4128-802E-41D311FFCE6F@freeswitch.org> <20110112153305.GF25773@hijacked.us> Message-ID: <20110112154925.GG25773@hijacked.us> Its also a LOT faster to clone: git clone git://git.freeswitch.org/freeswitch.git 11.51s user 3.64s system 5% cpu 4:45.63 total git clone git://github.com/FreeSWITCH/FreeSWITCH.git 11.53s user 3.42s system 29% cpu 51.069 total git clone https://github.com/FreeSWITCH/FreeSWITCH.git 13.29s user 3.87s system 29% cpu 57.786 total On top of that, it supports cloning over HTTP, which is helpful on some paranoid corporate networks (which is why I set it up in the first place, a friend couldn't clone FreeSWITCH from work). Andrew From msc at freeswitch.org Wed Jan 12 20:27:44 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 12 Jan 2011 17:27:44 +0000 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110112/0949f4e5/attachment.html From amonroy at auronix.com Wed Jan 12 21:45:03 2011 From: amonroy at auronix.com (Arturo Monroy) Date: Wed, 12 Jan 2011 12:45:03 -0600 Subject: [Freeswitch-dev] session:setHanhupHook is launched with session:tranfer Message-ID: <4D2DF6AF.6010404@auronix.com> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110112/d4f24d5e/attachment.html From m.sobkow at marketelsystems.com Thu Jan 13 00:31:50 2011 From: m.sobkow at marketelsystems.com (Mark Sobkow) Date: Wed, 12 Jan 2011 15:31:50 -0600 Subject: [Freeswitch-dev] Freeswitch timing out waiting for response from Erlang In-Reply-To: <4D2DF6AF.6010404@auronix.com> References: <4D2DF6AF.6010404@auronix.com> Message-ID: <4D2E1DC6.2070603@marketelsystems.com> I updated our Freeswitch built to today's git head, and am now having a problem with the Erlang-Freeswitch communications. Has the format of the freeswitch.xml configuration file changed at all lately? (Past couple of months) What's happening is we're getting Freeswitch starting and getting recognized by our freeswitch_monitor erlang process. I successfully registers the Freeswitch callbacks so that it should receive requests for config data from freeswitch. Freeswitch correctly requests the timezone configuration file, which we send back, logging the contents of the file we sent. However, 5 seconds later fs_cli reports that Freeswitch timed out waiting for a response, after which it makes no further requests for config data. This is code that's been working for over a year, hence my suspicion that something may have changed on the Freeswitch side of things. Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow at marketelsystems.com Web: http://www.marketelsystems.com Visit our Blog for industry related information. http://marketel-systems.blogspot.com/ From anthony.minessale at gmail.com Thu Jan 13 04:21:35 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 12 Jan 2011 19:21:35 -0600 Subject: [Freeswitch-dev] session:setHanhupHook is launched with session:tranfer In-Reply-To: <4D2DF6AF.6010404@auronix.com> References: <4D2DF6AF.6010404@auronix.com> Message-ID: you have to check if its transfer or hangup in the hook function On Wed, Jan 12, 2011 at 12:45 PM, Arturo Monroy wrote: > 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 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From bggoutham at gmail.com Fri Jan 14 15:36:55 2011 From: bggoutham at gmail.com (Goutham BG) Date: Fri, 14 Jan 2011 18:06:55 +0530 Subject: [Freeswitch-dev] Fwd: Peculiar behavior in FreeSWITCH when media streams for SRTP and RTP are offered in the same SDP In-Reply-To: References: Message-ID: 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 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 Hi, I am observing a peculiar behavior in FreeSWITCH-1.0.7 with SRTP. I have the following entry in my dialplan XML file: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110114/428ba2cc/attachment.html From brian at freeswitch.org Fri Jan 14 18:33:45 2011 From: brian at freeswitch.org (Brian West) Date: Fri, 14 Jan 2011 09:33:45 -0600 Subject: [Freeswitch-dev] Fwd: Peculiar behavior in FreeSWITCH when media streams for SRTP and RTP are offered in the same SDP In-Reply-To: References: Message-ID: Yes and if you read my reply you'll see what to do... you MUST set sip_secure_media=true before you answer the call otherwise it can't work. See default dialplan in default configs. /b On Jan 14, 2011, at 6:36 AM, Goutham BG wrote: > Sending the below query to FreeSWITCH-dev list as well. Any hints will be very helpful. > > Thanks > Goutham B G From steveayre at gmail.com Fri Jan 14 18:56:10 2011 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 14 Jan 2011 15:56:10 +0000 Subject: [Freeswitch-dev] Fwd: Peculiar behavior in FreeSWITCH when media streams for SRTP and RTP are offered in the same SDP In-Reply-To: References: Message-ID: his 1st message: 2 days ago his 2nd message: 3 hours ago your 1st reply: 21 minutes ago your 2nd reply: 20 minutes ago Not his fault Brian, looks like your mail server didn't get his 1st message in a timely manner. Regards, -Steve On 14 January 2011 15:33, Brian West wrote: > Yes and if you read my reply you'll see what to do... you MUST set > sip_secure_media=true before you answer the call otherwise it can't work. > > See default dialplan in default configs. > > /b > > On Jan 14, 2011, at 6:36 AM, Goutham BG wrote: > > > Sending the below query to FreeSWITCH-dev list as well. Any hints will be > very helpful. > > > > Thanks > > Goutham B G > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110114/f2d817b9/attachment.html From brian at freeswitch.org Fri Jan 14 19:08:13 2011 From: brian at freeswitch.org (Brian West) Date: Fri, 14 Jan 2011 10:08:13 -0600 Subject: [Freeswitch-dev] Fwd: Peculiar behavior in FreeSWITCH when media streams for SRTP and RTP are offered in the same SDP In-Reply-To: References: Message-ID: Seems my reply from yesterday didn't make it either thus I replied again and that one seems to have made it. :P I gotta see WTF is going on ... seems to have started doing this since we moved the mailing list server to a new IP. /b On Jan 14, 2011, at 9:56 AM, Steven Ayre wrote: > his 1st message: 2 days ago > his 2nd message: 3 hours ago > your 1st reply: 21 minutes ago > your 2nd reply: 20 minutes ago > > Not his fault Brian, looks like your mail server didn't get his 1st message in a timely manner. > > Regards, > -Steve > > > On 14 January 2011 15:33, Brian West wrote: > Yes and if you read my reply you'll see what to do... you MUST set sip_secure_media=true before you answer the call otherwise it can't work. > > See default dialplan in default configs. > > /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110114/56b06e9c/attachment-0001.html From brian at freeswitch.org Fri Jan 14 19:25:41 2011 From: brian at freeswitch.org (Brian West) Date: Fri, 14 Jan 2011 10:25:41 -0600 Subject: [Freeswitch-dev] FreeSWITCH 1.0.7 Message-ID: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org> http://latest.freeswitch.org/ Enjoy! /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110114/05e04b64/attachment.html From michelhabib at gmail.com Fri Jan 14 19:44:40 2011 From: michelhabib at gmail.com (Michel Habib) Date: Fri, 14 Jan 2011 18:44:40 +0200 Subject: [Freeswitch-dev] Processing Live Audio during a call between 2 extensions Message-ID: 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110114/955a127c/attachment.html From sbolshakov at altlinux.ru Fri Jan 14 23:08:58 2011 From: sbolshakov at altlinux.ru (Sergey Bolshakov) Date: Fri, 14 Jan 2011 23:08:58 +0300 Subject: [Freeswitch-dev] FreeSWITCH 1.0.7 References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org> Message-ID: >>>>> "Brian" == Brian West writes: > Enjoy! Could you please push version tag to git then ? Thanks. -- From dave at 3c.co.uk Sun Jan 16 03:48:24 2011 From: dave at 3c.co.uk (David Knell) Date: Sat, 15 Jan 2011 17:48:24 -0700 Subject: [Freeswitch-dev] Processing Live Audio during a call between 2 extensions In-Reply-To: References: Message-ID: <1295138904.2727.57.camel@dk-d820> Hi Michel, If it really needs to be an external application, then have a look at the docs (such as they are) for unicast here: http://wiki.freeswitch.org/wiki/Event_Socket#unicast Cheers -- Dave > 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. > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From mrene_lists at avgs.ca Mon Jan 17 06:12:36 2011 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sun, 16 Jan 2011 22:12:36 -0500 Subject: [Freeswitch-dev] [Freeswitch-users] Processing Live Audio during a call between 2 extensions In-Reply-To: References: Message-ID: <5AD8856E-3FF6-4EE3-9C6C-A68FAA6D4371@avgs.ca> Hi, Your best way is to make a freeswitch module and attach a media bug to the channel, you can take a look at mod_soundtouch's source, which sets one up (using switch_core_media_bug_add()) and changes the audio. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2011-01-14, at 11:44 AM, Michel Habib wrote: > 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. > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Mon Jan 17 18:23:51 2011 From: brian at freeswitch.org (Brian West) Date: Mon, 17 Jan 2011 09:23:51 -0600 Subject: [Freeswitch-dev] [Freeswitch-users] FreeSWITCH 1.0.7 In-Reply-To: <87ipxov50u.fsf@jdc.jasonjgw.net> References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org> <87ipxov50u.fsf@jdc.jasonjgw.net> Message-ID: <2013F94B-1AD7-44BD-85FE-6F6DF7753D3A@freeswitch.org> the odd numbers are not tagged in GIT. Even numbers are. Expect 1.0.8 in a month or so. /b On Jan 16, 2011, at 4:48 PM, Jason White wrote: > A minor point: the 1.0.7 release doesn't appear to have been tagged in > the Git repository. Perhaps the tag hasn't been pushed. git push --tags > should do it in that case. From tayeb.meftah at gmail.com Sun Jan 16 20:30:30 2011 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sun, 16 Jan 2011 18:30:30 +0100 Subject: [Freeswitch-dev] Processing Live Audio during a call between 2 extensions In-Reply-To: <1295138904.2727.57.camel@dk-d820> References: <1295138904.2727.57.camel@dk-d820> Message-ID: <4D332B36.6050304@gmail.com> david, i suggest you do it as a module will by better take a look to mod_spy that allready spy up on a specific uuid thanks Le 16/01/2011 01:48, David Knell a ?crit : > Hi Michel, > > If it really needs to be an external application, then have a look at > the docs (such as they are) for unicast here: > http://wiki.freeswitch.org/wiki/Event_Socket#unicast > > Cheers -- > > Dave > > >> 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. >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > -- Meftah Tayeb inum: +883510001288000 Phone: +13602276297 Fax: +12538020313 From karlowski.lukasz at gmail.com Mon Jan 17 20:22:39 2011 From: karlowski.lukasz at gmail.com (Lukasz Karlowski) Date: Mon, 17 Jan 2011 18:22:39 +0100 Subject: [Freeswitch-dev] Voiceglue support? Message-ID: Dear Freeswitch Team, I am writting to you to ask if you are planning to add support for Voiceglue? It is immportant to us(me and my academic group) to get know about it. So we will be very greatefull for your anwser. Your sincerelly, ?ukasz Kar?owski with his acedmic group. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110117/7e3da959/attachment.html From jmesquita at freeswitch.org Mon Jan 17 20:57:52 2011 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 17 Jan 2011 14:57:52 -0300 Subject: [Freeswitch-dev] Voiceglue support? In-Reply-To: References: Message-ID: I think that, before all, the licenses will have to be studied... The OpenVXI License is not something standard according to their license at: http://www.speech.cs.cmu.edu/openvxi/download.html So it seems that the so called ScanSoft is Nuance? Jo?o Mesquita 2011/1/17 Lukasz Karlowski > Dear Freeswitch Team, > I am writting to you to ask if you are planning to add support for > Voiceglue? > It is immportant to us(me and my academic group) to get know about it. > So we will be very greatefull for your anwser. > > Your sincerelly, > ?ukasz Kar?owski with his acedmic group. > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110117/4b088bc5/attachment.html From dujinfang at gmail.com Tue Jan 18 12:35:32 2011 From: dujinfang at gmail.com (Seven Du) Date: Tue, 18 Jan 2011 17:35:32 +0800 Subject: [Freeswitch-dev] relative_url_root in rails3 Message-ID: our whole arch depending on relative url to work. It turns unefficient in rails2, now they changed http://stackoverflow.com/questions/3181746/what-is-the-replacement-for-actioncontrollerbase-relative-url-root Please use scopy .. do in rails 3 ... we lost some flexibility between dev and production , but we could write a deploy script to substitute the routes named scope in the future. Anyway, hard code it for now. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From dujinfang at gmail.com Tue Jan 18 12:36:42 2011 From: dujinfang at gmail.com (Seven Du) Date: Tue, 18 Jan 2011 17:36:42 +0800 Subject: [Freeswitch-dev] relative_url_root in rails3 In-Reply-To: References: Message-ID: Sorry! ignore this I send to the wrong place... On Tue, Jan 18, 2011 at 5:35 PM, Seven Du wrote: > our whole arch depending on relative url to work. It turns unefficient > in rails2, now they changed > > http://stackoverflow.com/questions/3181746/what-is-the-replacement-for-actioncontrollerbase-relative-url-root > > Please use scopy .. do in rails 3 ... we lost some flexibility between > dev and production , but we could write a deploy script to substitute > the routes named scope in the future. Anyway, hard code it for now. > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj:? http://www.freeswitch.org.cn > -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From nospametor at gmail.com Tue Jan 18 20:35:15 2011 From: nospametor at gmail.com (M D) Date: Tue, 18 Jan 2011 12:35:15 -0500 Subject: [Freeswitch-dev] Freeswitch on ARM, Memory alignment issues Message-ID: Summary: Has anyone looked at/actively looking at the 32-bit boundary memory alignment requirement for the ARM architecture? Detail: I am running freeswitch on an ARM platform (dockstar based on Marvell Kirkwood). I have compiled and tested both freeswitch-1.0.6 and freeswitch-head(1,2), and found that stun lookups were failing with a "Timeout" error. Debugging the issue points to memory alignment issues specific to the ARM architecture. Basically, due to the fact that the uint8_t array "buf", is not aligned at a 32-bit memory boundary, the stun lookup packet sent to the stun server was malformed. Web searches confirm that a symptom of misaligned access on ARM is an apparent memory corruption. This can be fixed by forcing alignment of specific variables to 32-bit boundary. As an example, in the switch_stun_lookup function of file src/switch_stun.c , we currently have { switch_sockaddr_t *local_addr = NULL, *remote_addr = NULL, *from_addr = NULL; switch_socket_t *sock = NULL; uint8_t buf[260] = { 0 }; .... } If I change it to { switch_sockaddr_t *local_addr = NULL, *remote_addr = NULL, *from_addr = NULL; switch_socket_t *sock = NULL; uint8_t buf[260] __attribute__ (aligned(4)) = { 0 }; .... } or { switch_sockaddr_t *local_addr = NULL, *remote_addr = NULL, *from_addr = NULL; switch_socket_t *sock = NULL; uint8_t *buf; buf = (uint8_t*) malloc(sizeof(uint8_t)*260); .... } the stun lookup packet sent is correct.freeswitch is also able to parse the recvd packet and correctly determines the external IP. Random crashes coupled with looking at the code in switch_rtp suggests that the memory alignment issue may not be limited to this function or even this file. Has anyone looked at ARM memory alignment issues for freeswitch in general? cheers, -m (1) Current head leads to the situation where call gets connected but there is no-audio. I have not had the chance to debug this, but suspect this could be due to the "eat rtp .... " patch combined with ARM memory alignment issues. (2) The tested (audio working, some issues) git version is commit 715d250e171a94736b19019ac742f739899ad997 Author: Mathieu Parent Date: Wed Dec 15 21:29:52 2010 +0100 From mike at jerris.com Tue Jan 18 20:54:53 2011 From: mike at jerris.com (Michael Jerris) Date: Tue, 18 Jan 2011 12:54:53 -0500 Subject: [Freeswitch-dev] Freeswitch on ARM, Memory alignment issues In-Reply-To: References: Message-ID: take a look at swith_types.h 645-710. This is what you would need to do cross platform to align stuff. I am surprised the first way isn't aligned already, the first proposed change wont be cross platform, the second, if that works, should be fine (as long as you free it later of course) but I am unsure if that will actually work cross platform. The trick here is to figure out how to fix it once for all platforms. Mike On Jan 18, 2011, at 12:35 PM, M D wrote: > Summary: > Has anyone looked at/actively looking at the 32-bit boundary memory > alignment requirement for the ARM architecture? > > Detail: > I am running freeswitch on an ARM platform (dockstar based on Marvell > Kirkwood). I have compiled and tested both freeswitch-1.0.6 and > freeswitch-head(1,2), and found that stun lookups were failing with a > "Timeout" error. Debugging the issue points to memory alignment issues > specific to the ARM architecture. Basically, due to the fact that the > uint8_t array "buf", is not aligned at a 32-bit memory boundary, the > stun lookup packet sent to the stun server was malformed. Web searches > confirm that a symptom of misaligned access on ARM is an apparent > memory corruption. > > This can be fixed by forcing alignment of specific variables to 32-bit > boundary. As an example, in the switch_stun_lookup function of file > src/switch_stun.c , we currently have > > { > switch_sockaddr_t *local_addr = NULL, *remote_addr = NULL, > *from_addr = NULL; > switch_socket_t *sock = NULL; > uint8_t buf[260] = { 0 }; > .... > } > > If I change it to > { > switch_sockaddr_t *local_addr = NULL, *remote_addr = NULL, > *from_addr = NULL; > switch_socket_t *sock = NULL; > uint8_t buf[260] __attribute__ (aligned(4)) = { 0 }; > .... > } > > or > { > switch_sockaddr_t *local_addr = NULL, *remote_addr = NULL, > *from_addr = NULL; > switch_socket_t *sock = NULL; > uint8_t *buf; > buf = (uint8_t*) malloc(sizeof(uint8_t)*260); > .... > } > > the stun lookup packet sent is correct.freeswitch is also able to parse > the recvd packet and correctly determines the external IP. > > Random crashes coupled with looking at the code in switch_rtp suggests > that the memory alignment issue may not be limited to this function or > even this file. Has anyone looked at ARM memory alignment issues for > freeswitch in general? > > cheers, > -m > > (1) Current head leads to the situation where call gets > connected but there is no-audio. I have not had the chance to > debug this, but suspect this could be due to the "eat rtp .... " patch > combined with ARM memory alignment issues. > > (2) The tested (audio working, some issues) git version is > commit 715d250e171a94736b19019ac742f739899ad997 > Author: Mathieu Parent > Date: Wed Dec 15 21:29:52 2010 +0100 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org From nospametor at gmail.com Tue Jan 18 22:42:09 2011 From: nospametor at gmail.com (M D) Date: Tue, 18 Jan 2011 14:42:09 -0500 Subject: [Freeswitch-dev] Freeswitch on ARM, Memory alignment issues In-Reply-To: References: Message-ID: <20110118144209.5aa91f25@vaioz.no.net> On Tue, 18 Jan 2011 12:54:53 -0500 Michael Jerris wrote: In general, should I conclude that this issue has not been encountered before? > take a look at swith_types.h 645-710. This is what you would need to > do cross platform to align stuff. If I understood you correctly, you mean to point out a general cross platform issue and how it is dealt with withing freeswitch code. Right? Just to complete the info, the ARM processor is running Little-Endian. The code/settings/dialplan worked fine on a x86 linux machine, also little endian. Probably nothing new, but I thought I should fill in the details. > I am surprised the first way isn't > aligned already, the first proposed change wont be cross platform, > the second, if that works, should be fine (as long as you free it > later of course) but I am unsure if that will actually work cross > platform. The trick here is to figure out how to fix it once for all > platforms. > > Mike I expected that the original code would be aligned too. It just wasn't, atleast with gcc 4.3.3 + Code Sourcery enhancements. In gcc 4.4.5 ( I think, I don't have access to my development machine at the moment.) using the switch -falign-functions did align the buffer buf at a 32-bit boundary. Of course align-functions does not ensure that all uint8_t buffers would be aligned. Perhaps a conditional define would be more suitable fix, considering all platforms. As far as I can tell, the memory alignment requirement is ARM specific, which restricts this issue to gcc/linux. So in pseudo code #define ALIGN_ON_ARM ifdef GCC ifdef arm #undef ALIGN_ON_ARM #define ALIGN_ON_ARM __attribute__ (aligned(4)) #endif #endif and then in the source code uint8_t buf[260] ALIGN_ON_ARM = {0}; Would something along these lines work? The only issue is that it would touch a lot of the code. The use of structures with a uint8 buffer is likely widely used within freeswitch. -m > On Jan 18, 2011, at 12:35 PM, M D wrote: > > > Summary: > > Has anyone looked at/actively looking at the 32-bit boundary memory > > alignment requirement for the ARM architecture? > > From m.sobkow at marketelsystems.com Tue Jan 18 23:37:23 2011 From: m.sobkow at marketelsystems.com (Mark Sobkow) Date: Tue, 18 Jan 2011 14:37:23 -0600 Subject: [Freeswitch-dev] Before I submit a bug report, could someone take a quick look at this voicemail.conf? Message-ID: <4D35FA03.1040400@marketelsystems.com> Our last bug report on Erlang-Freeswitch integration turned out to be an XML error that wasn't reported by Freeswitch rather than a bug in the integration of the two. I was hoping someone could take a look at our response XML and confirm that we haven't got an XML error before I submit a bug report to JIRA. Right now it _appears_ that FS isn't processing the voicemail to email configuration response from Erlang, but as I said, we had a similar issue a few weeks ago that turned out to be an XML error. Thanks.
-- Mark Sobkow Senior Developer MarkeTel Multi-Line Dialing Systems LTD. 428 Victoria Ave Regina, SK S4N-0P6 Toll-Free: 800-289-8616-X533 Local: 306-359-6893-X533 Fax: 306-359-6879 Email: m.sobkow at marketelsystems.com Web: http://www.marketelsystems.com Visit our Blog for industry related information. http://marketel-systems.blogspot.com/ From mike at jerris.com Tue Jan 18 23:40:40 2011 From: mike at jerris.com (Michael Jerris) Date: Tue, 18 Jan 2011 15:40:40 -0500 Subject: [Freeswitch-dev] Freeswitch on ARM, Memory alignment issues In-Reply-To: <20110118144209.5aa91f25@vaioz.no.net> References: <20110118144209.5aa91f25@vaioz.no.net> Message-ID: <3E91CDF8-45EB-4805-99B0-0308B05AC6C4@jerris.com> On Jan 18, 2011, at 2:42 PM, M D wrote: > On Tue, 18 Jan 2011 12:54:53 -0500 > Michael Jerris wrote: > > In general, should I conclude that this issue has not been encountered > before? This has certainly been encountered before, on different platforms, including arm. I suspect this is a toolchain issue more than anything in this case as we have worked on arm for quite some time. > >> take a look at swith_types.h 645-710. This is what you would need to >> do cross platform to align stuff. > If I understood you correctly, you mean to point out a general cross > platform issue and how it is dealt with withing freeswitch code. Right? > Just to complete the info, the ARM processor is running Little-Endian. > The code/settings/dialplan worked fine on a x86 linux machine, also > little endian. Probably nothing new, but I thought I should fill in the > details. the trick is to fix it in the way that fixes it for the most things all at once. >> I am surprised the first way isn't >> aligned already, the first proposed change wont be cross platform, >> the second, if that works, should be fine (as long as you free it >> later of course) but I am unsure if that will actually work cross >> platform. The trick here is to figure out how to fix it once for all >> platforms. >> >> Mike > > > I expected that the original code would be aligned too. It just wasn't, > atleast with gcc 4.3.3 + Code Sourcery enhancements. In gcc 4.4.5 ( I > think, I don't have access to my development machine at the moment.) > using the switch -falign-functions did align the buffer buf at a 32-bit > boundary. Of course align-functions does not ensure that all uint8_t > buffers would be aligned. Is there a gcc arg that will make this work? If so we can test for that being available and use it when it is. > > Perhaps a conditional define would be more suitable fix, considering > all platforms. As far as I can tell, the memory alignment requirement > is ARM specific, which restricts this issue to gcc/linux. So in pseudo > code > > #define ALIGN_ON_ARM > ifdef GCC > ifdef arm > #undef ALIGN_ON_ARM > #define ALIGN_ON_ARM __attribute__ (aligned(4)) > #endif > #endif > > and then in the source code > > uint8_t buf[260] ALIGN_ON_ARM = {0}; > > Would something along these lines work? The only issue is that it would > touch a lot of the code. The use of structures with a uint8 buffer > is likely widely used within freeswitch. > I would much rather not litter the code with a ton of garbage like this, there should be a way to tell the compiler how we want alignment to work either with a single ifdef in a header, or with a compiler arg. > -m > >> On Jan 18, 2011, at 12:35 PM, M D wrote: >> >>> Summary: >>> Has anyone looked at/actively looking at the 32-bit boundary memory >>> alignment requirement for the ARM architecture? >>> > From msc at freeswitch.org Wed Jan 19 20:41:01 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 19 Jan 2011 09:41:01 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today Message-ID: Hello all! The agenda for today's conference call is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_01_19 We have Mitch Capper scheduled to speak to us on his experiences with embedding FreeSWITCH in other applications. In his case he created a FS-based softphone for Windows. We look forward to hearing what he's got! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110119/ba419857/attachment.html From nospametor at gmail.com Wed Jan 19 20:59:18 2011 From: nospametor at gmail.com (M D) Date: Wed, 19 Jan 2011 12:59:18 -0500 Subject: [Freeswitch-dev] Freeswitch on ARM, Memory alignment issues In-Reply-To: <3E91CDF8-45EB-4805-99B0-0308B05AC6C4@jerris.com> References: <20110118144209.5aa91f25@vaioz.no.net> <3E91CDF8-45EB-4805-99B0-0308B05AC6C4@jerris.com> Message-ID: <20110119125918.2196940e@vaioz.no.net> On Tue, 18 Jan 2011 15:40:40 -0500 Michael Jerris wrote: > > On Jan 18, 2011, at 2:42 PM, M D wrote: > > > On Tue, 18 Jan 2011 12:54:53 -0500 > > Michael Jerris wrote: > > > > In general, should I conclude that this issue has not been > > encountered before? > > This has certainly been encountered before, on different platforms, > including arm. I suspect this is a toolchain issue more than > anything in this case as we have worked on arm for quite some time. Would it be possible to know which tuple works? I.e. binutils,gcc, libc (glibc or uclibc). At the moment I am trying to build different combinations of the toolchain to see if I can isolate a working version. What is the best way to share a testcase with the list? > > I expected that the original code would be aligned too. It just > > wasn't, atleast with gcc 4.3.3 + Code Sourcery enhancements. In gcc > > 4.4.5 ( I think, I don't have access to my development machine at > > the moment.) using the switch -falign-functions did align the > > buffer buf at a 32-bit boundary. Of course align-functions does not > > ensure that all uint8_t buffers would be aligned. > > Is there a gcc arg that will make this work? If so we can test for > that being available and use it when it is. I could not find one. I think I understand why gcc would not align the uint8_t array to a 16/32-bit boundary: The alignment is data type dependent, and since uint8_t is 8-bits, there is no need to align it with a 32-bit boundary. For what it is worth, malloc documentation suggests that it will return memory suitably aligned for **any** variable type, which means that it should work correctly on all platforms. I am not sure if the run-time overhead would be prohibitively high. Thanks for your input on all this. -m From msc at freeswitch.org Wed Jan 19 22:15:25 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 19 Jan 2011 11:15:25 -0800 Subject: [Freeswitch-dev] Possible mod_smpp Message-ID: Hello all, If you are interested in giving monetary support to have a professional software firm create mod_smpp then please contact me off list and I will give you more details. Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110119/64be3c5b/attachment.html From nospametor at gmail.com Thu Jan 20 18:14:26 2011 From: nospametor at gmail.com (M D) Date: Thu, 20 Jan 2011 10:14:26 -0500 Subject: [Freeswitch-dev] Freeswitch on ARM, Memory alignment issues In-Reply-To: <3E91CDF8-45EB-4805-99B0-0308B05AC6C4@jerris.com> References: <20110118144209.5aa91f25@vaioz.no.net> <3E91CDF8-45EB-4805-99B0-0308B05AC6C4@jerris.com> Message-ID: <20110120101426.605b0651@vaioz.no.net> On Tue, 18 Jan 2011 15:40:40 -0500 Michael Jerris wrote: > > On Jan 18, 2011, at 2:42 PM, M D wrote: > > > On Tue, 18 Jan 2011 12:54:53 -0500 > > Michael Jerris wrote: > > > > In general, should I conclude that this issue has not been > > encountered before? > > This has certainly been encountered before, on different platforms, > including arm. I suspect this is a toolchain issue more than > anything in this case as we have worked on arm for quite some time. > Sorry for the noise. Looks like a gcc-4.3 issue. I have tested the following code with gcc-4.3.3, gcc-4.3.5 and gcc-4.4.4 and gcc-4.4.5. gcc-4.4.4 comes from gentoo crossdev environment, everything else is from openwrt build system. The 4.4 versions align the buf2 in the code below with the 32-bit boundary while the 4.3 versions don't. All test cases compiled with -march=armv5te. No other compiler flags/switches used. Optimization options -Os vs -O2 do not seem to make a difference. If anyone has ready access to arm-gcc and an arm machine, I would be curious to see my experience is consistent with others'. cheers, -m P.S Code from memory. ----- test_align.c ---- #include #include #include #include /* typedef struct { uint16_t type; uint16_t length; char id[16]; } switch_stun_packet_header_t; */ void main(void) { uint8_t buf1[260] = {0}; char j; uint8_t buf2[260] = {0}; //switch_stun_packet_header_t *header; printf("buf1 starts at %x\n",buf1); printf("j starts at %x\n",&j); printf("buf2 starts at %x\n",buf2); } ------------------------------ From edpimentl at gmail.com Sun Jan 23 18:59:36 2011 From: edpimentl at gmail.com (EdPimentl) Date: Sun, 23 Jan 2011 10:59:36 -0500 Subject: [Freeswitch-dev] [Freeswitch-users] Bounty Offered for A2Billing equivalent for FreeSWITCH In-Reply-To: References: Message-ID: ASTPP already support FreeSwitch. Review the features missing you want to implement. If you decide ASTPP is a viable solution, I would contribute to the bounty. http://www.astpp.org/?q=node/165 -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110123/03fa3498/attachment.html From dujinfang at gmail.com Mon Jan 24 09:18:31 2011 From: dujinfang at gmail.com (Seven Du) Date: Mon, 24 Jan 2011 14:18:31 +0800 Subject: [Freeswitch-dev] email and reverse DNS Message-ID: Hi US-Team, We couldn't send email to some services like AOL. They need reverse DNS to work. So question: 1) How do we send email in the original eleutian system? If I'm right it directly send via google. So is there any limit on sending via google? how to set that? 2) We had reverse DNS on softlayer but as we are going to kill those servers, if 1) is not available we may need to ask reverse DNS from TCT. So, should we setup a centra mail server? 3) There current setup is to go directly from each server. Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From dujinfang at gmail.com Mon Jan 24 09:35:33 2011 From: dujinfang at gmail.com (Seven Du) Date: Mon, 24 Jan 2011 14:35:33 +0800 Subject: [Freeswitch-dev] email and reverse DNS In-Reply-To: References: Message-ID: sorry again. please ignore this. I have another dev list to send to but choosed the wrong address. Will take more care about this. Thanks. On Mon, Jan 24, 2011 at 2:18 PM, Seven Du wrote: > Hi US-Team, > > We couldn't ?send email to some services like AOL. They need reverse > DNS to work. So question: > > 1) How do we send email in the original eleutian system? If I'm right > it directly send via google. So is there any limit on sending via > google? how to set that? > > 2) We had reverse DNS on softlayer but as we are going to kill those > servers, if 1) is not available we may need to ask reverse DNS from > TCT. So, should we setup a centra mail server? > > 3) There current setup is to go directly from each server. > > Thanks. > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj:? http://www.freeswitch.org.cn > -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From prashant.lamba at gmail.com Mon Jan 24 10:06:08 2011 From: prashant.lamba at gmail.com (Prashant Lamba) Date: Mon, 24 Jan 2011 12:36:08 +0530 Subject: [Freeswitch-dev] Voiceglue support? In-Reply-To: References: Message-ID: > 2011/1/17 Jo?o Mesquita > > I think that, before all, the licenses will have to be studied... The > OpenVXI License is not something standard according to their > license at:?http://www.speech.cs.cmu.edu/openvxi/download.html > So it seems that the so called ScanSoft is Nuance? I know that from SpeechWorks, it went to Vocalocity. I think now its being managed by Commetrex under a GPL license. From cbeeton at avaya.com Tue Jan 25 00:05:34 2011 From: cbeeton at avaya.com (Beeton, Carolyn (Carolyn)) Date: Mon, 24 Jan 2011 16:05:34 -0500 Subject: [Freeswitch-dev] Display update after consult transfer to bridge/loopback app Message-ID: I have an application defined in the dialplan, accessed through a "bridge" to a loopback socket application so that consult transfer to it works correctly. If I dial it directly on a Polycom set, the set displays "Outbound Call/async full". I can overcome this by exporting the callee_id_name/number variables in the dialplan. However, if I transfer a set to this application via consult transfer, the set ends up with a "Outbound Call/app=socket:..." display and I can't figure out how to get it to show the name/number of the application. Any ideas? I'm running on today's git latest (6294bc1620c). Thanks, Carolyn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110124/a5e3dcd5/attachment.html From anthony.minessale at gmail.com Tue Jan 25 00:39:17 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 24 Jan 2011 15:39:17 -0600 Subject: [Freeswitch-dev] Display update after consult transfer to bridge/loopback app In-Reply-To: References: Message-ID: {origination_caller_id_name=foo,origination_caller_id_number=1234} in the outbound dial string. On Mon, Jan 24, 2011 at 3:05 PM, Beeton, Carolyn (Carolyn) wrote: > I have an application defined in the dialplan, accessed through a "bridge" > to a loopback socket application so that consult transfer to it works > correctly.? If I dial it directly on a Polycom set, the set displays > "Outbound Call/async full".? I can overcome this by exporting the > callee_id_name/number variables in the dialplan.? However, if I transfer a > set to this application via consult transfer, the set ends up with a > "Outbound Call/app=socket:..." display and I can't figure out how to get it > to show the name/number of the application.? Any ideas?? I'm running on > today's git latest (6294bc1620c). > > Thanks, > Carolyn > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From anthony.minessale at gmail.com Tue Jan 25 00:39:53 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 24 Jan 2011 15:39:53 -0600 Subject: [Freeswitch-dev] Display update after consult transfer to bridge/loopback app In-Reply-To: References: Message-ID: d'oh I mean {origination_callee_id_name=foo,origination_callee_id_number=1234} On Mon, Jan 24, 2011 at 3:39 PM, Anthony Minessale wrote: > {origination_caller_id_name=foo,origination_caller_id_number=1234} in > the outbound dial string. > > > On Mon, Jan 24, 2011 at 3:05 PM, Beeton, Carolyn (Carolyn) > wrote: >> I have an application defined in the dialplan, accessed through a "bridge" >> to a loopback socket application so that consult transfer to it works >> correctly.? If I dial it directly on a Polycom set, the set displays >> "Outbound Call/async full".? I can overcome this by exporting the >> callee_id_name/number variables in the dialplan.? However, if I transfer a >> set to this application via consult transfer, the set ends up with a >> "Outbound Call/app=socket:..." display and I can't figure out how to get it >> to show the name/number of the application.? Any ideas?? I'm running on >> today's git latest (6294bc1620c). >> >> Thanks, >> Carolyn >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From cbeeton at avaya.com Tue Jan 25 17:18:08 2011 From: cbeeton at avaya.com (Beeton, Carolyn (Carolyn)) Date: Tue, 25 Jan 2011 09:18:08 -0500 Subject: [Freeswitch-dev] Display update after consult transfer to bridge/loopback app In-Reply-To: References: Message-ID: Thank you, that fixes it. Carolyn > -----Original Message----- > From: freeswitch-dev-bounces at lists.freeswitch.org > [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On > Behalf Of Anthony Minessale > Sent: Monday, January 24, 2011 4:40 PM > To: freeswitch-dev at lists.freeswitch.org > Subject: Re: [Freeswitch-dev] Display update after consult > transfer to bridge/loopback app > > d'oh > > I mean > {origination_callee_id_name=foo,origination_callee_id_number=1234} > > > On Mon, Jan 24, 2011 at 3:39 PM, Anthony Minessale > wrote: > > > {origination_caller_id_name=foo,origination_caller_id_number=1234} in > > the outbound dial string. > > > > > > On Mon, Jan 24, 2011 at 3:05 PM, Beeton, Carolyn (Carolyn) > > wrote: > >> I have an application defined in the dialplan, accessed > through a "bridge" > >> to a loopback socket application so that consult transfer > to it works > >> correctly.? If I dial it directly on a Polycom set, the > set displays > >> "Outbound Call/async full".? I can overcome this by > exporting > >> the callee_id_name/number variables in the dialplan.? > However, if I > >> transfer a set to this application via consult transfer, > the set ends > >> up with a "Outbound Call/app=socket:..." display and I > can't figure > >> out how to get it to show the name/number of the application.? Any > >> ideas?? I'm running on today's git latest (6294bc1620c). > >> > >> Thanks, > >> Carolyn > >> _______________________________________________ > >> FreeSWITCH-dev mailing list > >> FreeSWITCH-dev at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-de > >> v > >> http://www.freeswitch.org > >> > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > From juanjo at comellas.org Wed Jan 26 00:48:22 2011 From: juanjo at comellas.org (Juan Jose Comellas) Date: Tue, 25 Jan 2011 18:48:22 -0300 Subject: [Freeswitch-dev] [Freeswitch-users] FreeSWITCH 1.0.7 In-Reply-To: <2013F94B-1AD7-44BD-85FE-6F6DF7753D3A@freeswitch.org> References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org> <87ipxov50u.fsf@jdc.jasonjgw.net> <2013F94B-1AD7-44BD-85FE-6F6DF7753D3A@freeswitch.org> Message-ID: Is there any commit ID by which we can identify what v1.0.7 is? Or is v1.0.7. whatever is on the master branch until v1.0.8 is released? On Mon, Jan 17, 2011 at 12:23 PM, Brian West wrote: > the odd numbers are not tagged in GIT. Even numbers are. Expect 1.0.8 in > a month or so. > > /b > > On Jan 16, 2011, at 4:48 PM, Jason White wrote: > > > A minor point: the 1.0.7 release doesn't appear to have been tagged in > > the Git repository. Perhaps the tag hasn't been pushed. git push --tags > > should do it in that case. > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110125/a3765b28/attachment.html From anthony.minessale at gmail.com Wed Jan 26 01:04:49 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 25 Jan 2011 16:04:49 -0600 Subject: [Freeswitch-dev] [Freeswitch-users] FreeSWITCH 1.0.7 In-Reply-To: References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org> <87ipxov50u.fsf@jdc.jasonjgw.net> <2013F94B-1AD7-44BD-85FE-6F6DF7753D3A@freeswitch.org> Message-ID: The odd number releases are nightly builds. Once we start rolling them, it means we are testing for release of the next major version. So the whole GIT tree is all considered 1.0.7 no matter what state its in. The next major release will be 1.2.0 at which time a stable branch will be created. On Tue, Jan 25, 2011 at 3:48 PM, Juan Jose Comellas wrote: > Is there any commit ID by which we can identify what v1.0.7 is? Or is > v1.0.7. whatever is on the master branch until v1.0.8 is released? > > On Mon, Jan 17, 2011 at 12:23 PM, Brian West wrote: >> >> the odd numbers are not tagged in GIT. ?Even numbers are. ?Expect 1.0.8 in >> a month or so. >> >> /b >> >> On Jan 16, 2011, at 4:48 PM, Jason White wrote: >> >> > A minor point: the 1.0.7 release doesn't appear to have been tagged in >> > the Git repository. Perhaps the tag hasn't been pushed. git push --tags >> > should do it in that case. >> >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From steveayre at gmail.com Wed Jan 26 01:34:06 2011 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 25 Jan 2011 22:34:06 +0000 Subject: [Freeswitch-dev] [Freeswitch-users] FreeSWITCH 1.0.7 In-Reply-To: References: <5612B825-2DC9-493E-A77C-75EF1F77AE06@freeswitch.org> <87ipxov50u.fsf@jdc.jasonjgw.net> <2013F94B-1AD7-44BD-85FE-6F6DF7753D3A@freeswitch.org> Message-ID: 1.0.7 is git head, built daily, the release candidate for 1.0.8 Steve on iPhone On 25 Jan 2011, at 21:48, Juan Jose Comellas wrote: > Is there any commit ID by which we can identify what v1.0.7 is? Or is v1.0.7. whatever is on the master branch until v1.0.8 is released? > > > On Mon, Jan 17, 2011 at 12:23 PM, Brian West wrote: > the odd numbers are not tagged in GIT. Even numbers are. Expect 1.0.8 in a month or so. > > /b > > On Jan 16, 2011, at 4:48 PM, Jason White wrote: > > > A minor point: the 1.0.7 release doesn't appear to have been tagged in > > the Git repository. Perhaps the tag hasn't been pushed. git push --tags > > should do it in that case. > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110125/6745d83a/attachment.html From msc at freeswitch.org Wed Jan 26 03:30:33 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 25 Jan 2011 16:30:33 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Wednesday Jan 26 - Vestec Speech Recognition Message-ID: Hello! I wanted to let everyone know that we have special guests on the conference call this Wednesday: Kashif Kahn and Dr. Jin-Myung Won, both from Vestec. Dr. Won has been deeply involved in ASR development and will be making himself available to answer your technical questions. Please join the conference call and be part of this great discussion about advanced ASR technology! -Michael Agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2011_01_26 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110125/32b2e0aa/attachment.html From msc at freeswitch.org Wed Jan 26 21:07:27 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 26 Jan 2011 10:07:27 -0800 Subject: [Freeswitch-dev] *FreeSWITCH Conf Call Now! Hear About New Vestec Speech Rec Engine! Message-ID: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110126/a4055a06/attachment.html From gilles.depatie at mapleworks.com Wed Jan 26 23:24:33 2011 From: gilles.depatie at mapleworks.com (Gilles Depatie) Date: Wed, 26 Jan 2011 15:24:33 -0500 Subject: [Freeswitch-dev] switch_xml_config.c Message-ID: <2146B97617A64E0985101460B7E0FBC4@mapleworks.com> Hi I'm running into what appears to be a questionable range interpretation issue in switch_xml_config_parse_event function. When enforce_min and enforce_max are TRUE should the value not be inclusive of MIN and MAX? Say a value range of min = '0' and max = '1000', should values of 0 and 1000 not be acceptable? The current statement: if ((int_options->enforce_min && !(intval > int_options->min)) || (int_options->enforce_max && !(intval < int_options->max))) excludes those values presently The error message seems to imply that the range is inclusive of the min and max values I've had a look at the latest git version or the file and in this respect nothing has changed from the version I am currently using GILLES Gilles Depatie, Senior S/W Designer, MapleWorks Technology O: 819-776-6066 x353 | gilles.depatie at mapleworks.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110126/f4e4090b/attachment-0001.html From bernhard.suttner at winet.ch Thu Jan 27 03:09:26 2011 From: bernhard.suttner at winet.ch (Bernhard Suttner) Date: Thu, 27 Jan 2011 01:09:26 +0100 Subject: [Freeswitch-dev] fsctl pause Message-ID: <20110127010926.a8da18c0@mail.winet.ch> Hi, if I use "fsctl pause" freeswitch should not handle any new call but a INVITE will still be send answered from FreeSWITCH with 407 Proxy Authentication Required. This message go back to the peer. If the peer then send the INVITE with credentials (with authentication) FreeSWITCH sends bacck the 503 "No more sessions" because of the "pause state". Shouldn?t freeswitch respond with 503 for the first INVITE and not only for the second INVITE with authentication? Best regards, Bernhard From anthony.minessale at gmail.com Thu Jan 27 03:16:32 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 26 Jan 2011 18:16:32 -0600 Subject: [Freeswitch-dev] fsctl pause In-Reply-To: <20110127010926.a8da18c0@mail.winet.ch> References: <20110127010926.a8da18c0@mail.winet.ch> Message-ID: possibly, we'd have to do something similar line 6263 in sofia.c over in line 840 before we challenge. when event is nua_i_invite On Wed, Jan 26, 2011 at 6:09 PM, Bernhard Suttner wrote: > Hi, > > if I use "fsctl pause" freeswitch should not handle any new call but a INVITE will still be send answered from FreeSWITCH with 407 Proxy Authentication Required. This message go back to the peer. If the peer then send the INVITE with credentials (with authentication) FreeSWITCH sends bacck the 503 "No more sessions" because of the "pause state". > > Shouldn?t freeswitch respond with 503 for the first INVITE and not only for the second INVITE with authentication? > > Best regards, > Bernhard > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From bernhard.suttner at winet.ch Thu Jan 27 13:03:15 2011 From: bernhard.suttner at winet.ch (Bernhard Suttner) Date: Thu, 27 Jan 2011 11:03:15 +0100 Subject: [Freeswitch-dev] fsctl pause In-Reply-To: References: <20110127010926.a8da18c0@mail.winet.ch> Message-ID: <6ff05a41-3521-4ac9-a8eb-d2fa113ce0c0@winet.ch> Hi, I have created a Jira ticket (FS-3012) and attached a patch which should fix this issue. Please let me know, what you think about that! Best regards, Bernhard -----Urspr?ngliche Nachricht----- Von: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale Gesendet: Donnerstag, 27. Januar 2011 01:17 An: freeswitch-dev at lists.freeswitch.org Betreff: Re: [Freeswitch-dev] fsctl pause possibly, we'd have to do something similar line 6263 in sofia.c over in line 840 before we challenge. when event is nua_i_invite On Wed, Jan 26, 2011 at 6:09 PM, Bernhard Suttner wrote: > Hi, > > if I use "fsctl pause" freeswitch should not handle any new call but a INVITE will still be send answered from FreeSWITCH with 407 Proxy Authentication Required. This message go back to the peer. If the peer then send the INVITE with credentials (with authentication) FreeSWITCH sends bacck the 503 "No more sessions" because of the "pause state". > > Shouldn?t freeswitch respond with 503 for the first INVITE and not only for the second INVITE with authentication? > > Best regards, > Bernhard > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _______________________________________________ FreeSWITCH-dev mailing list FreeSWITCH-dev at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org From nospametor at gmail.com Fri Jan 28 18:47:55 2011 From: nospametor at gmail.com (M D) Date: Fri, 28 Jan 2011 10:47:55 -0500 Subject: [Freeswitch-dev] Freeswitch on ARM, Memory alignment issues In-Reply-To: <3E91CDF8-45EB-4805-99B0-0308B05AC6C4@jerris.com> References: <20110118144209.5aa91f25@vaioz.no.net> <3E91CDF8-45EB-4805-99B0-0308B05AC6C4@jerris.com> Message-ID: Dear Michael, On Tue, Jan 18, 2011 at 3:40 PM, Michael Jerris wrote: > > On Jan 18, 2011, at 2:42 PM, M D wrote: > >> On Tue, 18 Jan 2011 12:54:53 -0500 >> Michael Jerris wrote: >> >> In general, should I conclude that this issue has not been encountered >> before? > > This has certainly been encountered before, on different platforms, including arm. ?I suspect this is a toolchain issue more than anything in this case as we have worked on arm for quite some time. I have done some further testing and it appears that the issue is not necessarily a toolchain issue. It is true that some toolchains seem to exacerbate the issue, but even with gcc-4.4.5 some memory alignment problems seem to be still there. The linux kernel has an alignment trap on ARM cpus. It is possible to activate alignment fix-ups at the kernel level by executing echo 2 > /proc/cpu/alignment Restarting freeswitch after the alignment trap is active shows cat /proc/cpu/alignment User: 25068 System: 1 Skipped: 0 Half: 28 Word: 0 DWord: 25041 Multi: 0 User faults: 2 (fixup) My tests indicate that majority of the alignment traps are triggered from freeswitch code. I would really appreciate if anyone can provide any information on successful builds on ARM. Information regarding the build system used would be invaluable in isolating the cause. The page at http://lecs.cs.ucla.edu/wiki/index.php/XScale_alignment has some general details regarding memory alignment on ARM. -m From singhujjwal at gmail.com Fri Jan 28 10:22:05 2011 From: singhujjwal at gmail.com (Ujjwal SIngh) Date: Fri, 28 Jan 2011 12:52:05 +0530 Subject: [Freeswitch-dev] offering two "m=" lines from FreeSWITCH Message-ID: Hi, I am new to FreeSWITCH, I was just wondering that can we initiate Best Effort SRTP offer from FreeSWITCH in the same way as Polycom phones do, i.e sending two "m=" lines for the same media, the first line having a SAVP profile, while the second one having the normal AVP profile. Kindly please let me know how can i do this. Thanks in advance. Regards, Ujjwal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110128/a72d6a66/attachment.html From williams.owen at gmail.com Fri Jan 28 14:31:02 2011 From: williams.owen at gmail.com (Owen Williams) Date: Fri, 28 Jan 2011 11:31:02 +0000 Subject: [Freeswitch-dev] SIP INFO carrying MSML Message-ID: Hi all, I'm looking into the possibility of getting basic msml support into freeswitch. We have a few applications hosted on a remote SIP Application server that already use MSML against another media server. The application acts as a back2back between a caller and the media server and sends info messages (inside an invite dialog) to the media server with the INFO sdp containing MSML (e.g. play file, join conf, etc). I'm a bit new to freeswitch but I'm guessing mod_sofia would need to modified to send info events to a new module or to the ESL, and mod_sofia also needs to be capable of receiving events to send INFOs back to the remote B2B. There was a bounty completed about 18 months ago (Generate and be notified of SIP INFO messages) however this may have been only for out of dialog INFOs. http://wiki.freeswitch.org/wiki/Bounty#Generate_and_be_notified_of_SIP_INFO_messages Any information as to whether the above bounty made it into trunk and how it can be enabled would be appreciated. I'd like to have a play with writing a MSML processor through ESL as I've far more experience with ESL than writing c modules. Regards, Owen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110128/15d1df5a/attachment.html From jerry.richards at teotech.com Mon Jan 31 23:07:00 2011 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 31 Jan 2011 12:07:00 -0800 Subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F2C52EF@VA3DIAXVS351.RED001.local> Hello All, When I set the call_timeout variable (to 12 seconds) before a bridge to an outbound PSTN call, Freeswitch always rings for 24 seconds before going to voicemail. If I call an internal extension, then it uses the call_timeout variable. Do you know why it would do this? Here is the line that sets the call_timeout variable: switch_channel_set_variable(channel, "call_timeout", "6"); Here is the PSTN bridge log: EXECUTE sofia/internal/2003 at 192.168.72.79:5060 bridge({presence_id=2001 at 192.168.72.79}[lcr_carrier=Carrier / Location 1/INTERNAL PRI TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_prid/a/15248142341 at g1) Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20110131/c05d5566/attachment.html From anthony.minessale at gmail.com Mon Jan 31 23:13:19 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 31 Jan 2011 14:13:19 -0600 Subject: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F2C52EF@VA3DIAXVS351.RED001.local> References: <2BF7FB90DF25EA4485949F3AF2B9D6960B3F2C52EF@VA3DIAXVS351.RED001.local> Message-ID: You should use {originate_timeout=N} but that may not be the actual problem. You would probably have to look at a trace with debug level etc to be sure. On Mon, Jan 31, 2011 at 2:07 PM, Jerry Richards wrote: > Hello All, > > > > When I set the call_timeout variable (to 12 seconds) before a bridge to an > outbound PSTN call, Freeswitch always rings for 24 seconds before going to > voicemail.? If I call an internal extension, then it uses the call_timeout > variable.? Do you know why it would do this? > > > > Here is the line that sets the call_timeout variable: > > switch_channel_set_variable(channel, "call_timeout", "6"); > > > > Here is the PSTN bridge log: > > EXECUTE sofia/internal/2003 at 192.168.72.79:5060 > bridge({presence_id=2001 at 192.168.72.79}[lcr_carrier=Carrier / Location > 1/INTERNAL PRI > TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg_prid/a/15248142341 at g1) > > > > Thanks, > > Jerry > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900