April 2012 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Conference recording pause/resume problem
by Mariusz Czulada 13 Jul '12

13 Jul '12

Hi, I try to prepare a patch which allows to pause/resume a given recording. I'm in the middle of the road: pause is working, by resume - not :(. I'd like to ask for your help in finding a problem. I'm not sure if I handle member and file flags in the thread sub correctly. BTW. I lost some time playing with member-fnode until found fh in conference_record_thread_run is a proper handler. What is the conference_file_node_t member->fnode used for, if not for handling a recording file? Regards, Mariusz

2 5

[Freeswitch-dev] questions about making a new endpoint
by Seven Du 29 Jun '12

29 Jun '12

Hi, I'm working on a mod_rtsp endpoint the idea is to stream sip video to a rtsp client such as VLC and also a sip client could "call" a rtsp camera later. I copied some codes from mod_sofia and it is too long to understand for me so I'd like to get some inputs either highlight me how to do or point out where to read from mod_sofia. 1) VLC "calls" in as a-leg and bridge to a sip client with video(or play_fsv), I made it sort of working now, but - How to disable the read_frame and read_video frame io_routine ? I set them to NULL and it fails on read, so I made a fake read function there, kind of ugly. - I hard coded to H264 and I can see video on VLC, I'd like to do like sip later negotiation then when b-leg sip client offers SDP and offer to a-leg when DESCRIBE - looks like switch_io complains when no audio codec, is it possible to do video only calls? or is it possible to set audio and video rtp to a different path ? 2) Call a rtsp camera would be easy later 3) Maybe a client can call in and record? I don't have such a client. I would like to share the code later if others interested on this. Thanks, Seven. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

4 5

[Freeswitch-dev] ClueCon 2012 Preview: Vestec Automatic Speech Recognition
by Michael Collins 30 Apr '12

30 Apr '12

Greetings! This week we have a small taste of ClueCon 2012: The Vestec <http://www.vestec.com?cc120430>team will be joining us on the FreeSWITCH community conference<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12-0430>call this Wednesday, May 2. Vestec produces high-quality automatic speech recognition (ASR) software for open source telephony projects like Asterisk and FreeSWITCH. We look forward to having a Vestec engineer on the call with us this week to give us a technical look at their product and to answer community questions. Vestec will also be at ClueCon 2012 as a Silver Sponsor and have committed to being available for one-on-one consulting for those wanting to discuss their specific projects. Yet another reason to join us at ClueCon this August! The ClueCon 2012 speakers list is starting to take shape. We have a number of "fan-favorites" scheduled as well as a few new surprises. We are pleased to have returning for a third consecutive year Mr. Philip Zimmermann, inventor of PGP <http://en.wikipedia.org/wiki/PGP> and designer of the the ZRTP <http://zfone.com?cc12-0430> secure VoIP protocol. As we've previously announced <http://freeswitch.org/node/388?cc12-0430>, ZRTP is now included in the FreeSWITCH distribution. Philip will be discussing VoIP security in general, as well developments with the ZRTP project. Register <http://www.cluecon.com/register?cc120430> for ClueCon 2012 today and take advantage of ClueCon Power Play, which gives you extra entries in the world famous giveaways we have each year. Feel free to contact us if you have any questions about attending ClueCon, sponsoring, or speaking at the event. We look forward to seeing you there - in less than 100 days! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0430

1 0

[Freeswitch-dev] ClueCon 2012 Preview: Vestec Automatic Speech Recognition
by Michael Collins 30 Apr '12

30 Apr '12

Greetings! This week we have a small taste of ClueCon 2012: The Vestec <http://www.vestec.com?cc120430>team will be joining us on the FreeSWITCH community conference<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12-0430>call this Wednesday, May 2. Vestec produces high-quality automatic speech recognition (ASR) software for open source telephony projects like Asterisk and FreeSWITCH. We look forward to having a Vestec engineer on the call with us this week to give us a technical look at their product and to answer community questions. Vestec will also be at ClueCon 2012 as a Silver Sponsor and have committed to being available for one-on-one consulting for those wanting to discuss their specific projects. Yet another reason to join us at ClueCon this August! The ClueCon 2012 speakers list is starting to take shape. We have a number of "fan-favorites" schedule as well as a few new surprises. We are pleased to have returning for a third consecutive year Mr. Philip Zimmermann, inventor of PGP <http://en.wikipedia.org/wiki/PGP> and designer of the the ZRTP <http://zfone.com?cc12-0430> secure VoIP protocol. As we've previously announced, ZRTP is now included in the FreeSWITCH distribution. Philip will be discussing VoIP security in general, as well developments with the ZRTP project. Register <http://www.cluecon.com/register?cc120430> for ClueCon 2012 today and take advantage of ClueCon Power Play, which gives you extra entries in the world famous giveaways we have each year. Feel free to contact us if you have any questions about attending ClueCon, sponsoring, or speaking at the event. We look forward to seeing you there - in less than 100 days! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0430

1 0

[Freeswitch-dev] newbie build problem: libapr configure hangs on pthreads detection
by Christian Wiedmann 27 Apr '12

27 Apr '12

I'm new to FreeSWITCH and am in the process of trying to build the system on Ubunto 11.10. I grabbed the source using git. The problem I ran into is that configure would hang when detecting "CFLAGS needed for pthreads" in libs/apr. When I did a ps after trying a couple of times, I discovered a bunch of defunct conftest processes. I added a pthread_join() to the conftest code used for the detection and the configure was able to complete. My question is whether this behavior indicates an incompatibility or missing library on my system. Should I be worried about future issues on this system? Thanks, -Christian

1 0

[Freeswitch-dev] mod_isac questions
by Steve Richardson 26 Apr '12

26 Apr '12

Hello all, I've been trying to work with the reasonably new mod_isac over the last couple of days, with no luck. We are launching a smartphone app as a companion to our hardware product (which is based on FreeSWITCH). The app is basically a really simplified SIP client that supports G.722 and iSAC for wideband audio calls. I can make G.722 calls into FreeSWITCH and they work fine. Unfortunately, I haven't been as lucky with iSAC. I have a few different failure scenarios, all of which tested with the latest git this morning -- FreeSWITCH Version 1.1.beta1 (git-7a147e4 2012-04-25 17-14-55 -0500): Scenario 1: With my FS codec-prefs set to "ISAC,G722,...", the remote end offers up "a=rtpmap:96 ISAC/16000" in the INVITE. When I answer the call by bridging it to a conference, FS unilaterally decides that "a=rtpmap:96 ISAC/32000" is more to its liking in the SDP. Though FS thinks it has answered the call, the remote end stays ringing (playing ringback), and no connection ever actually seems to happen. I have to individually hang up each end of the call; if I just hang up one end, the call remains in some half-active state. Scenario 2: If I change the codec-prefs to "ISAC@16000h", things go a little bit better in terms of the codec negotiation, but the call immediately drops after this happens: 2012-04-27 01:27:39.490028 [NOTICE] mod_conference.c:6228 Channel [sofia/external-4/Steve(a)comrex2.onsip.com] has been answered 2012-04-27 01:27:39.490028 [DEBUG] sofia.c:5602 Channel sofia/external-4/Steve(a)comrex2.onsip.com entering state [completed][200] 2012-04-27 01:27:39.490028 [DEBUG] mod_conference.c:6111 Raw Codec Activation Success L16@16000hz 1 channel 60ms 2012-04-27 01:27:39.490028 [WARNING] switch_core_codec.c:691 Codec L16 Exists but not at the desired implementation. 48000hz 60ms 2012-04-27 01:27:39.490028 [DEBUG] mod_conference.c:6159 Raw Codec Activation Failed L16@48000hz 1 channel 60ms 2012-04-27 01:27:39.490028 [WARNING] switch_core_codec.c:789 Codec is not initialized! The conference is set up at 48kHz/20ms, if that matters any. It has a mod_portaudio participant in it, also running at 48kHz/20ms. Scenario 3: If I change the codec-prefs to "ISAC@16000h@60i", things behave exactly as scenario #2, leading me to think that's what the default value is when the "@##i" suffix is left off. Scenario 4: If I change the codec-prefs to "ISAC@16000h@30i", things work marginally 'better,' but still aren't working properly. In that case, the call is established, but I can get no audio in the FS->remote direction, and the audio in the other direction is badly mangled. It sounds like periodic short frames of working audio, interleaved with silence frames, and the delay seems to spool out more and more over a very short period of time. "Choppy" is the layman description I would use. So, a few questions: - Given that iSAC is an adaptive codec that can place one or two frames (30ms or 60ms) of audio data into an RTP payload, how does this play with FS seeming to assume a fixed frame size? - What was used to test mod_isac when it was written? There are not a lot of soft phones out there that I've found that support the iSAC codec, so it's hard to come up with a known-good reference for testing. - Any other advice for things to test or try out? thanks, Steve -- *Stephen Richardson Senior Software Engineer* Comrex Corporation 19 Pine Rd Devens MA 01434 USA v 978.784.1764 f 978.784.1717 steve(a)comrex.com <mailto:steve@comrex.com> http://www.comrex.com/

1 0

[Freeswitch-dev] Fwd: embed freeswitch
by Xijing Dai 26 Apr '12

26 Apr '12

hey, I am newbie to the freeswitch. I am wondering if it's possible to use extension modules after I embeded freeswitch into my app. Anyone? Cheers

2 2

[Freeswitch-dev] Reminder: FreeSWITCH Conference Call Today
by Michael Collins 25 Apr '12

25 Apr '12

Just reminding everyone of our call today: http://wiki.freeswitch.org/wiki/FS_weekly_2012_04_25 We will be discussing the execution of dialplan apps and APIs using the "execute_on" and "api_on" channel variables. We'll also have a lively discussion about why shouldn't do billing stuff inline and why people do it anyway. :) See you at 1PM Eastern, 10AM Pacific -Michael

1 0

[Freeswitch-dev] FreeSWITCH News and Notes
by Michael Collins 23 Apr '12

23 Apr '12

Greetings everyone. We have several items for your consideration this week: - The FreeSWITCH Community Conference Call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_04_25>will take place at 1PM Eastern, 10AM Pacific this coming Wednesday April 25. We will be having an informal discussion about executing dialplan applications or APIs using the "execute_on" and "api_on" family of channel variables. We look forward to having you there with us. - The authors of the FreeSWITCH Cookbook will be taking part in an interview on the weekly VUC call <http://www.voipusersconference.org/>this Friday April 27. - Vestec still has some ASR starter kits to give away to FreeSWITCH users around the world. If you have need of speech recognition in your telephony application then contact marketing(a)cluecon.com and ask for more information. Vestec already supports languages in 10 of the largest economies of the world and more languages are in development. Please make plans to support both conference calls this week. If you've never been to either one I highly recommend checking them out. They are very interesting. In fact, they are a small taste of what ClueCon is like: hanging out with fellow telecom enthusiasts and having technical discussions. It really is a lot of fun. Speaking of ClueCon, be sure to register <http://www.cluecon.com/register/> today. Don't forget that if you register in April you get extra entries in the giveaway drawings we have throughout the conference. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

1 0

[Freeswitch-dev] Test message, please ignore
by Michael Collins 23 Apr '12

23 Apr '12

1 0