August 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Unreliable DTMF
by Alex Lake 13 Sep '13

13 Sep '13

I've got an application that doesn't seem to work well with DTMF coming in (supposedly using rfc2833) from some phones - notably Samsung S3s. I have told my carrier to use rfc2833. I have put dtmf-type="rfc2833" into my SIP profile. I see "a=fmtp:101 0-16" in the session progress message I was wondering what this message "Activate Buggy RFC2833 Mode" means! When a normal user hits a DTMF key, we get this in the log: 2013-07-25 13:26:19.308267 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 1:744 2013-07-25 13:26:19.308267 [DEBUG] switch_channel.c:471 RECV DTMF 1:744 Does this all indicate that rfc2833 is working?! Anyone else had trouble with S3s?

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IM messaging on FreeSwitch
by mehroz 10 Sep '13

10 Sep '13

Hi, Anyone with an experience of SIP SIMPLE IM Messaging implemention? I dont see the stuff well documented on http://wiki.freeswitch.org/wiki/Mod_sms <http://wiki.freeswitch.org/wiki/Mod_sms> I followed simple steps to test it with reply back as in default chatplan. <?xml version="1.0" encoding="utf-8"?> <include> <context name="default"> <extension name="demo"> <condition field="to" expression="^(.*)$"> <action application="reply" data="Hello, you said: ${_body}"/> </condition> </extension> </context> </include> I tested it with eyeBeam, Jitsi, it doesnt work, and i do not see reply back. and i see in logs: 2013-03-07 14:58:49.090936 [INFO] mod_sms.c:334 Processing text message 9999->9999 in context default Chatplan: 9999 parsing [default->demo] continue=false Chatplan: 9999(a)198.84.61.52 Regex (PASS) [demo] to(9999(a)198.84.61.52) =~ /^(.*)$/ break=on-false Chatplan: 9999(a)198.84.61.52 Action reply(Hello, you said: ${_body}) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/IM-messaging-on-FreeSwitch-tp… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Thanks to all those that make ClueCon Possible!!
by Ken Rice 08 Sep '13

08 Sep '13

ClueCon 2013 was a smashing success and you had everything to do with it! We were very glad to have you with us this year and we hope you enjoyed it every bit as much as we did. The ClueCon team is currently working on getting all of the videos and presentation slides put up on the ClueCon website. More information will be sent out soon. In the meantime be sure to join us each Wednesday at 1PM Eastern time for the ClueCon Weekly conference call. What about ClueCon 2014? It's already in the works! 2014 marks the 10th anniversary of ClueCon - a milestone you'll not want to miss. We will be putting out a call for speakers as well as getting our sponsors arranged in the next few months. Stay tuned for more information. If you missed Michael Giagnocavo's ClueCon 2013 presentation, "Easy Real-Time CDR Reporting with VoltDB," check it out here <http://goo.gl/EywbDm> . And then grab a free trial of VoltDB <http://go.voltdb.com/voltdb-enterprise-software-download> the database that helps Michael manage high-volume CDRs. The ClueCon team would like to thank the following sponsors: Nenad, Moises, and all the good people at Sangoma <http://www.sangoma.com> . Sangoma has been with us since the very first ClueCon. Sean, Jeff, and the folks at Flowroute <http://www.flowroute.com> . Arnie and the gang at NACT <http://www.nact.com> . Kashif and the team over at Vestec <http://www.vestec.com> . Norm and the VoiceNetwork <http://www.voicenetwork.ca> team. Venky and the good folks at Plivo <http://www.plivo.com> . The whole OrecX <http://www.orecx.com> team. John, Nicole and the rest of the gang at OnSIP <http://www.onsip.com> . Darren and all the techies at 2600hz <http://www.2600hz.com> . John, Dave, and all the rest of the Truphone <http://www.truphone.com> guys. Kevin and the team at Ooma <http://www.ooma.com> . Tommy and the folks at Snom <http://www.snom.com> . The tokbox <http://www.tokbox.com> team. The NG Communications <http://www.ngcommunications.com> team. We hope to see you all again in 2014! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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eavesdrop not sending video
by D D 08 Sep '13

08 Sep '13

Hi, I tried to use eavesdrop to capture the video of a parked call, but I only receive the audio. I have followed the directions described at http://wiki.freeswitch.org/wiki/Video-recording (including a rebuild after doing the git checkout of video-media-bug), and I can successfully use record_fsv to record the video to a file. Any ideas why the eavesdrop does not send the video? Thanks, David

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FreeSWITCH Core - Cassandra Database
by Lloyd Aloysius 07 Sep '13

07 Sep '13

Hi All Does any one have experience with FreeSWITCH Core and Cassandra Database. Sounds like Cassandra (http://cassandra.apache.org/) have a better replication system. This will help to distribute freeswitch across multiple data centers and provide a better fail over system. Any feedback ? Thank you in advance. Thanks Lloyd

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mod_opus codec on Windows
by Jeff Leung 05 Sep '13

05 Sep '13

As I heard on last Wednesday's conference call, mod_opus isn't "available" on Windows, basically meaning that FreeSWITCH users can't fully implement WebRTC on Windows Platforms just yet. I have taken the steps to have the codec successfully build on this platform; However it's a pain in the neck to build as the opus library generates multiple libs which includes CELT and SILK codecs (Go figure). Shoot me a message off-list if you would like to use the binary I've built for Win64 platforms. Before anybody asks for a 32-bit binary, the short answer is going to be no. I won't be providing a 32-bit binary as I intend this binary to be used in a production environment.

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Google voice call fails
by Terry Barnum 04 Sep '13

04 Sep '13

I've been reading the freeswitch book (great read!) and playing with my first freeswitch install on a Mac mini host with freeswitch installed via git into a Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones and a Yealink T32G, internally between extensions, using a free DID to test incoming calls and a free but limited iptel account to test outbound. My next step was to try and play with real outbound calls so I followed these instructions for Google voice: <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just PCMU but no change. xmpp appears to be connecting and authenticating to google correctly. The log of an attempted call (edited for privacy) is at <http://pastebin.freeswitch.org/21338> Snippets of the log that looks suspicious to a freeswitch noob: 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543(a)voice.google.com Where should I be looking to try and fix? Freeswitch is behind the firewall with a private IP but STUN says it's working (or at least it looks that way to me). Thanks, -Terry

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Lua script problem
by Phil Quesinberry 01 Sep '13

01 Sep '13

Hi everyone, Would anyone have any ideas as to why the following statement would not work and end up assigning a null value to scripts_dir? scripts_dir = string.sub(debug.getinfo(1).source,2,string.len(debug.getinfo(1).source)-(st ring.len(argv[0])+1)); I'm pretty sure that this is related to a much larger problem that we're experiencing where MWI has stopped working completely on all extensions. We started experiencing intermittent problems with this after making recent updates to FS and FusionPBX before it stopped working entirely. Marc Crane, the developer of FusionPBX, is under the impression that this is either an FS bug or that something has changed with FS' handling of MWI. I don't know lua so I can't weigh in on this myself. I filed a Jira on this, see the following for more details, including the full script the above excerpt is taken from: http://jira.freeswitch.org/browse/FS-5739 Any input would be greatly appreciated. Phil Quesinberry Q Systems Engineering, Inc. (410) 969-8002 http://www.qsystemsengineering.com <http://www.qsystemsengineering.com/>

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inbound-bypass-media for 1.2 (stable version)
by Nikolay Rogoshchenkov 30 Aug '13

30 Aug '13

Is it possible to use this option <param name="inbound-bypass-media" value="true"/> for 1.2 version? -- Rogoshchenkov Nikolay

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Suggestion for additional syntax for "continue_on_fail"
by Nathan Neulinger 30 Aug '13

30 Aug '13

It seems like there are a bunch of codes that would typically indicate a problem with a gateway, as opposed to a valid return - like no answer or busy. My current list that I'm working with that I'm treating as "upstream failed to handle it for some reason" is: CALL_REJECTED,GATEWAY_DOWN,RECOVERY_ON_TIMER_EXPIRE,NORMAL_TEMPORARY_FAILURE,DESTINATION_OUT_OF_ORDER, NORMAL_UNSPECIFIED,UNALLOCATED_NUMBER,NO_ROUTE_DESTINATION It would be nice to be able to both use negation type syntax, as well as 'collection's. I was thinking something like this: ALL,!USER_BUSY,!NO_ANSWER Example from OpenSSL: SSLCipherSuite ALL:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:NULL-MD5:NULL-SHA For now, I'm just using a variable for it defined centrally. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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