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Use FreeSWITCH to solve a problem and win a prize!
by Anthony Minessale 30 Jun '17

30 Jun '17

This year at the ClueCon Coder Games there is a category for the most innovative use of FreeSWITCH to solve a problem. The prize is a bar top arcade. Get started now! Check out the other challenges and prizes on our web site: http://cluecon.com/coder-games.html -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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Issue with Sofia SIP: username and colons
by Lorenzo Miniero 22 Jun '17

22 Jun '17

Hi, I'm not sure this is the right place to discuss a potential issue related to Sofia SIP, especially considering I'm going to describe something that happens in another software, but I thought that, considering that Freeswitch uses Sofia SIP as well and has kept it alive so far, it would be of interest to you guys as well, as you may be affected too. To give you some context, I'm using Sofia SIP in one of the plugins of my WebRTC server, Janus. Specifically, this plugin acts as a SIP client on behalf of WebRTC users, meaning I create NUA instances for them, doing registers, invites and stuff like that from the server side and exposing a simpler API and events (not SIP) to users. This usually works fine, but I recently encountered a weird problem that can occur when users try to REGISTER and provide a username with a colon in it. This fails, and apparently the cause is in how the NUTAG_AUTH works. As you probably know, NUTAG_AUTH is the tag nua_authenticate requires when answering a REGISTER challenge, and it needs a string that is basically a colon-separated concatenation of digest/realm/user/pass, e.g.: Digest:"nokia proxy":xyz:secret This breaks when the username contains a colon, as the method that parses this concatenated string (auc_credentials() in auth_client.c) seems to do a "brutal" split on the colon character. I tried wrapping the username in quotes, but that didn't work, which seems to confirm what the code suggests. Apparently the only way to get Sofia SIP to digest the username with the colon in it, is escaping the colon (%3A) in there, but then this obviously results in a broken authentication when it gets to the registrar, as the usernames differ (ciccio:123 != ciccio%3A123). Some may argue that it should be the registrar's responsibility to unescape the username part, but this is a quoted string in the authorization challenge and not the username part of a URI, so I don't think they're required to do that. Is this something you encountered as well? Being more of a SIP server, I guess this may be seen as less of an issue in Freeswitch, but it might still occur if/when you're doing active REGISTERs yourself. If it's a known problem, do you happen to know of any workaround that wouldn't require patching auc_credentials to do the right thing? (which might be either changing the delimiter in NUTAG_AUTH to a much less used character, or automatically unescaping %3A internally when formatting the username). Thanks in advance for any feedback you may have for me! Lorenzo

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how to link two audio conference in freeswitch?
by Dbsk Dbsk 19 Jun '17

19 Jun '17

hi guys, i just want to know how to do that in freeswitch. for example, i have two audio conference 3001 and 3002, which has it's own members in conference ,and now i want all user in the two conference can hear all the talking. first it may think transfer the member of 3001 to 3002, but it is not want i want, cause if i may link any two conference, and when i cancel such link, the member in the original conferece was not affected. thanks for any clue in advance. dbsk

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CDR Tests
by Paul Zillmann 12 Jun '17

12 Jun '17

Hello, currently I'm working on a revival of mod_cdr_mongodb and I'm looking for (unit) tests for CDR modules in general and / or tests for switch_ivr_generate_json_cdr() in switch_ivr.c. I've ported switch_ivr_generate_json_cdr from cJSON to libbson. Thanks, Paul Zillmann

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Answering call using inbound event socket mod
by Baljit Kaur 10 Jun '17

10 Jun '17

Hello Team, I am a newbie to FreeSWITCH.I am trying to make an IVR application in C#. I want to use inbound mode for answering an incoming call to the app and then play menu options and place a call to the another number. currently, my app is connected successfully to FreeSWITCH, receiving events but no playing the .wav file against the incoming call. how would this be possible? here is my code while (eslConnection.Connected() == ESL_SUCCESS) { eslEvent = eslConnection.RecvEvent(); eslConnection.Execute("answer", null, strUuid); eslConnection.Execute("playback", "C:\\FS_GIT\\Win32\\Debug\\sounds\\music\\8000\\suite-espanola-op-47-leyenda.wav", strUuid); Console.WriteLine(string.Format("{0}:Inbound CallId:{1}", eslEvent.getType(), strUuid)); Console.WriteLine(eslEvent.Serialize(String.Empty)); }

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Hey Devs What Single Board computers are you playing with
by Ken Rice 05 Jun '17

05 Jun '17

Hey guys, Quick Query, What Single Board Computers like the RPi, BeagleBone etc are you playing with for Communications and IoT stuffs? Ken

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Shared line/call appearance, barging, and hangups
by Ilia Mirkin 17 May '17

17 May '17

Hi Anthony, I've been checking out the SLA functionality in FreeSWITCH. Briefly, the issue I'm running into is that if a second station barges in on the first station, and then the first station hangs up, then the call completely hangs up. I'm trying to make something where the call keeps going, effectively replicating the old analog experience. In commit a511ff3026b8, some short 5 years ago, you implemented barging as eavesdropping. You also added code, which has largely remained unchanged, in sofia_handle_sip_i_bye which looks like it attempts to deal with the situation I describe. However it's not working. With some hacks I was able to get the second call to not hang up, but the media does not pass through to it anymore. The core issue is that other_uuid is null (which is the SWITCH_SIGNAL_BOND_VARIABLE). It may be due to the fact that I was testing this by dialing e.g. a conference application or echo application, and then joined the call - as such the original call wasn't bonded in the first place. However I think I also tested it with something more akin to a regular call and it also failed. I'm less sure of that though. Something else that I think I've observed is that if the initial station puts the call on hold, then eavesdropping (i.e. barging) station is effectively muted as well. Again, primarily testing with e.g. the echo application. Any advice you can give on these issues would be greatly appreciated! Thanks, -ilia

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How audio mixing works in Freeswitch conference?
by kaiduan xie 15 May '17

15 May '17

Hi, Can anyone explain how audio mixing works in Freeswitch audio conference and where is the source code? Thanks a lot in advance, /Kaiduan

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Help with autotools
by Alberto Mijares 15 May '17

15 May '17

Hi guys, I'm currently working, with a co-worker, in a pkgsrc package for FreeSwitch 1.6.17. We are writing the Makefile from scratch because we think that many patches from previous versions are no longer needed in this version. We are working on FreeBSD 11. None of us are savvy in development nor know the details of how autotools work. However, we were able to identify: 1) The distribution file from files.freeswitch.org differs from the git version, and is easier to fetch via http in pkgsrc; is the preferred method of getting distfiles. 2) There's a bootstrap.sh or rebootstrap.sh (depending on the source of the distribution) that uses libtool to generate the configure scripts. From this .sh script we have a better understanding of the build process. If you have all the dependencies and run this script, everything goes fine. However, we are not very clear on if we should run this .sh script from pkgsrc framework of if the pkgsrc framework should be able to run the libtool utility for itself in the same way the script does. pkgsrc has some vars USE_LIBTOOL= yes USE_TOOLS+= automake autoconf gmake perl pkg-config so those tools are installed in the build process as dependencies. What I can't identify is if pkgsrc knows how to run them accordingly or those vars are useful only for having preinstalled the tools and using them within the script. I also noted that some directories were removed from sources in git and not in distfile. Is there any chance to make them equal? Thanks in advance for your guidance. Best regards, Alberto Mijares

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❤an unexpected surprise
by Mohammed Habib 01 May '17

01 May '17

Hello, I just wanted to share with you my feelings about an unexpected surprise my friends made former)) It was so cool, just take a look http://download49.thebooziere.com Regards, Mohammed Habib Sent from Mail for Windows 10

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