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

[Freeswitch-dev] FreeSWITCH Conference Call Tomorrow (Wed Feb 22)
by Michael Collins 21 Feb '12

21 Feb '12

Hey all! Yes, I promise that I will come up for air and hang out with the community tomorrow. :) We have a special guest joining us tomorrow: Stefan Wintermeyer from Amooma. He will be talking about Gemeinschaft v4 and all the interesting things that lead to going with FreeSWITCH instead of Asterisk. The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_02_22 Look forward to speaking with you all tomorrow! -Michael

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[Freeswitch-dev] Call For Community Assistance
by Michael Collins 21 Feb '12

21 Feb '12

Hello all! As you may have noticed, I've been rather absent from the IRC channel and mailing lists for the past week or so. Like others on the FreeSWITCH and CudaTel team I have learned first hand just how valuable Brian is and how much work he produces. We are all trying to pitch in and lighten the load. To that end I wanted to say thank you to Ken Rice, Jeff Lenk, Avi Marcus, and all the rest of the community members who have been stepping up to help out on IRC and the mailing list. We definitely can use more help in this regard. I'd like to ask everyone subscribed to this mailing list to review the recent threads and look for questions which you can answer quickly and then do so. If there are any subjects that you cannot answer right away but could research, I'd ask that you send a reply to the thread indicating your willingness to do some research on the subject at hand. If everyone can do just a little bit then we'll get a lot accomplished. One other thing you can help with: testing the new sounds batch. I've rolled sounds version 1.0.18 and put it up on files.freeswitch.org. I've run it on several of my test machines and now I'd like to have a lot of others do so as well. Just be sure to do this on a test machine before trying it on a production machine. :D Edit fs_src/build/sounds_version.txt and set the en-callie version to "1.0.18" Save the file, then issue "make cd-sounds-install" It should download the latest sound tarballs Once the downloads are done do some basic testing like calling in to a conference, sending and retrieving voicemails, etc. Let me know if you run into any issues. Thanks again for all your help! -Michael

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[Freeswitch-dev] conference play / stop / PAUSE/RESUME
by Gledhill, James 20 Feb '12

20 Feb '12

I have been trying to do the same thing as was asked in an earlier question, but it was asked 10 months ago and I did not want to use the same thread, as it went a different direction than my question. I understood the concept of the answer, but I am not quite sure how to implement it. I would like to make a call to another person (via FS), be able to record parts of the call, and then play it back to both participants - being able to use the features of uuid_fileman (pause, seek +/-, etc), then record another part of the conversation, playback, etc. >From other searches, it was made clear that this had to be done with a conference call. ===== From the previous thread Apr 18, 2011; 11:30am Re: conference play / stop / PAUSE/RESUME ==== Anthony Minessale II - wrote if you need this level of granularity, I suggest instead of the conference play stuff that is built in, that you use the originate command to call a new channel into the conference to play the file then use the uuid_fileman api command to manipulate the playback. On Sat, Apr 16, 2011 at 1:33 PM, Dale Trub <[hidden email]> wrote: > We let the conference host play a file to the conference and want to give > them a control to "pause" such a file, and then resume it. > Has anyone found a way to do this? > Would someone create something like that for a modest bounty? We could > incorp in the code we use, as well as submit to the project overall. > -Dale ======= End Previous thread quoted ========= Does anyone have an example of this? I would like to implement it is either dailplan or LUA, but Javascript or anything else would be good examples. I have tried a number of different ways and have not gotten it to work. * I may not be doing "use the originate command to call a new channel into the conference to play the file" properly. Suggestion on this? One thing I tried was in LUA. Session1 is the original call-ER, Session2 is the call-ED party - both sessions are (1&2) are uuid_transer-ed to a conference bridge. * api:execute("uuid_transfer", session1:get_uuid() .. " conf_11_111 XML callRecPlayback" ); * api:execute("uuid_transfer", session2:get_uuid() .. " conf_11_111 XML callRecPlayback" ); * * session3 = freeswitch.Session("loopback/conf_11_111/callRecPlayback/XML"); * session3:setAutoHangup(false); * freeswitch.consoleLog("info","\n\n SEssion 3 == ".. session3:get_uuid() .." \n\n"); * * session:execute("bind_digit_action", "cool,444,api:uuid_broadcast," .. session3:get_uuid() .. " ivr/ivr-welcome_to_freeswitch.wav both"); The bind_digit_action works and calls the api, but fs_cli reports CHANNEL_KILL on the loopback channel, and I do not hear anything 2012-02-20 10:19:09.381934 [DEBUG] switch_rtp.c:3405 RTP RECV DTMF 4:640 2012-02-20 10:19:09.681942 [DEBUG] switch_rtp.c:3405 RTP RECV DTMF 4:640 2012-02-20 10:19:09.981952 [DEBUG] switch_rtp.c:3405 RTP RECV DTMF 4:1120 2012-02-20 10:19:09.981952 [DEBUG] mod_dptools.c:185 sofia/callRecPlayback/1004@172.30.110.112:5083 Digit match binding [api:uuid_broadcast][8d48d97a-5c74-4104-bc66-e424a9a1d155 ivr/ivr-welcome_to_freeswitch.wav both] 2012-02-20 10:19:09.981952 [DEBUG] switch_core_session.c:1012 Send signal loopback/conf_11_111-a [BREAK] 2012-02-20 10:19:09.981952 [DEBUG] mod_loopback.c:475 loopback/conf_11_111-a CHANNEL KILL 2012-02-20 10:19:09.981952 [DEBUG] mod_dptools.c:244 sofia/callRecPlayback/1004@172.30.110.112:5083 Digit match binding [api:uuid_broadcast][8d48d97a-5c74-4104-bc66-e424a9a1d155 ivr/ivr-welcome_to_freeswitch.wav both] api executed, +OK Message Sent

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[Freeswitch-dev] How to understand codebase
by Arif Hossain 16 Feb '12

16 Feb '12

I'm quite new to freeswitch. I have some experience in asterisk. But Mostly about how to configure asterisk. The codebase of freeswitch is quite large and seems overhelming at first sight. I'm a moderately good c programmer but i don't have a lot of experience understanding a huge codebase only by reading code. I saw freeswitch does not come with a "developer manual" or something like that. Which would be very helpful. Because i do have a moderate knowledge about sip and telephony technologies. So any other ways to understand freeswitch codebase(the core, and important modules like media session handling etc) efficiently? I'm willing to "learn" and contribute to the project. I know "knowledge" is itself a "resource". But any help regarding this would be wonderful. At this time i have checked out freeswitch code from git, loaded it in eclipse, and trying to understand some core data structures. But I've realized it would be very difficult to understand everything by myself. Thanks in advance. -- "You have a voice" ----The King's Speech Public Key : $ gpg --keyserver keyserver.ubuntu.com --recv-key C88CFC23

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[Freeswitch-dev] radius_cdr
by Miha Zoubek 16 Feb '12

16 Feb '12

Hi, just one information about radius_cdr modul. For A leg it sends stop&start packet and also same for B leg. I have also one another softswitch that behaviour is just like on FS (stop&start for A and B leg). The difference that I have noticed is that FS for A leg sends different session ID that for B leg. So, this mean that I will have two entries in my sql table for one call. Other softswith send same session ID for a and b leg, so this mean one entry in sql table for one call. So, what is right? Is it possible to change this behaviour on FS? Thanks! -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o.

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[Freeswitch-dev] FreeSWITCH Conference Call Today: 2600Hz Team
by Michael Collins 15 Feb '12

15 Feb '12

Hello all! Apologies for the late notice. When Brian West isn't at my beck and call I just can't function. :) Today's conference call will start shortly. Here's the agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2012_02_15 We are looking forward to having Darren Schreiber (IRC: pyite) and the rest of the 2600Hz team come talk to us about the state of their projects. -MC

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[Freeswitch-dev] mod_nibblebill
by Erjan Tylepov 15 Feb '12

15 Feb '12

Hi all, I'm using mod_nibblebill, in which there is a line of code "if (profile->times->answered < 1) {", that determines whether a call has been answered. The abovesaid worked fine with a commit 883de14f3068d8cc269cccce3895effc750ef108. But after pulling a commit 303de52bad2d43efcfe41520e4afbae919cf4864, the module doesn't work as before, namely, after a call has been answered the field "times->answered" in a structure "switch_caller_profile" is still 0. Does anybody know what has changed? Regards, Yerzhan.

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[Freeswitch-dev] backslash in call-id
by Seven Du 14 Feb '12

14 Feb '12

Hi, I have a sip client which generates call-id with backslashes which causing error: ERR: [insert into sip_registrations ............. values ('<c)X/Iz\' I looked into http://sofia-sip.sourceforge.net/refdocs/sip/group__sip__call__id.html looks like backslash '\' is valid in call-id so is it ok to to let switch_mprintf %q also quote backslash ? should I report a jira? Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

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[Freeswitch-dev] Increasing Packetization time for g729 codec
by Arif Hossain 14 Feb '12

14 Feb '12

I have tested that the packetization time (ptime) for g729 codec does go beyond 50ms. I want to modify the code so freeswitch can repacketize the rtp stream upto 200 ms. What i'm trying to achieve is : client-----150ms------>FREESWITCH------20m-------->IPTDM Terminations AND IPTDM termination------20ms---->FREESWITCH-------->200ms--->client. First one can be done through configuration. But second one is not yet possible for my test environment. I'm using sangoma codec card. It supports packetization time upto 200ms. So please help me with some pointers about the freeswitch codebase to achieve this. After modifying i will submit the patches. I have not filed any bug report yet. I'm going to do that tod

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[Freeswitch-dev] switch_ivr_bridge.c question
by teldev 11 Feb '12

11 Feb '12

Re-worded my original post a bit, sorry if it becomes a new thread...having Nabble issues. In switch_ivr_bridge.c on line 1474, we have the following lines of code: 1474 switch_channel_clear_state_handler(originator_channel, NULL); 1475 switch_channel_clear_state_handler(originatee_channel, NULL); 1476 switch_channel_set_state_flag(originator_channel, CF_BRIDGE_ORIGINATOR); 1477 switch_channel_clear_flag_recursive(originatee_channel, CF_BRIDGE_ORIGINATOR); 1478 switch_channel_add_state_handler(originator_channel, &uuid_bridge_state_handlers); 1479 switch_channel_add_state_handler(originatee_channel, &uuid_bridge_state_handlers); Is the call to “switch_channel_set_state_flag” a typo? Since line 1476 triggers an asynchronous flag change, and line 1477 changes the same flag synchronously, on line 1478 the value of the flag becomes unpredictable since the asynchronous change from line 1476 can happen before or after line 1477 (mostly after, of course). Thanks again. -- View this message in context: http://freeswitch-dev.4355.n6.nabble.com/switch-ivr-bridge-c-question-tp435… Sent from the FreeSWITCH-dev mailing list archive at Nabble.com.

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