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by Андрей Аппалонов 11 Sep '14

11 Sep '14

Hello! Tell me please, can FreeSwitch capture RTP packets without the other applications? If so, how is this done? Best Regards, Appalonov Andrei.

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-=>@ FreeSWITCH News and Notes - Weekly Bug Hunt @<=-
by Ken Rice 04 Sep '14

04 Sep '14

Greetings FreeSWITCHers! Just a reminder we did send out a survey to all the ClueCon attendees from SurveyMonkey. If you received this request to do the survey and have not already completed it, please do so! This will give us the feedback we need to make ClueCon better next year! If you have open Jiras, Patches, Pull Requests or related things and need help or want to help others with these, join us on 888 today at 2PM Central for the FreeSWITCH Weekly BugHunt! How do you join? Visit https://conference.freeswitch.org/ with your WebRTC enabled browser and click the Call 888 Button! Have a headset handy! Need to call in via the PSTN, SIP or other method? See https://confluence.freeswitch.org/display/FREESWITCH/ClueCon+Weekly+Conferen ce+call for more ways to call in! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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error building latest FS with branch v1.4
by Nuno Reis 04 Sep '14

04 Sep '14

Hi Travis. I've tried to build the latest v1.4 today and found out that your commit: "Remove dead assignments in `switch.c`" (aa9bbf82) is breaking the build, at least on CentOS 6 x86_64. src/switch.c: In function 'check_fd': src/switch.c:254: error: ignoring return value of 'write', declared with attribute warn_unused_result src/switch.c: In function 'main': src/switch.c:1192: error: ignoring return value of 'read', declared with attribute warn_unused_result At top level: cc1: error: unrecognized command line option "-Wno-unused-result" make[2]: *** [src/freeswitch-switch.o] Error 1 make[2]: Leaving directory `/home/devel/rpmbuild/BUILD/freeswitch-wavecom' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/home/devel/rpmbuild/BUILD/freeswitch-wavecom' make: *** [all] Error 2 error: Bad exit status from /var/tmp/rpm-tmp.ThFmYG (%build) I've just undo the commit again and the build has been successful. Cheers, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature] <http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

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RECOVERY_ON_TIMER_EXPIRE + Changing Hangup Cause
by Massimo Varriale 03 Sep '14

03 Sep '14

Hi All! I've submitted my question to the User List but actually this could be something more tricky and of course could involve some dev guru out there... This is my first question...and what question... I'm a skilled developer and I'm almost familiar with telephony in general since it's 15 years experience in telco companies. However I'm still facing a problem I couldn't find any solution yet and i hope someone could help in finding a solution As my present company can't afford an expensive soft switch I'm using Freeswitch 1.4 with a simple dialplan.xml and some scripts to load CDRs into mysql to send incoming calls to my remote peers. At present the live traffic is only 1 incoming IP and 1 outgoing peer, quite easy. On some calls I'm getting "408 Request Timeout" and FS is giving me a RECOVERY_ON_TIMER_EXPIRE cause (IDSN 102). The problem is that as most of the switches are not able to make automatic rerouting based on this cause and so, I will like to use instead the ISDN Cause 18 (I suppose "480 Temporarily unavailable"). Is this possible? I've used this snippet of code that use another relcause but with same concept (taken as is and pasted into the dialplan) but doesn't work as the 408 is given after 183. <extension name="AllDestinations_Inbound"> <condition field="destination_number" expression="^"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=19"/> <action application="bridge" data="sofia/gateway/remote_outbound_peer/${destination_number}"/> <action application="transfer" data="408to503"/> </condition> </extension>  <extension name="408to503"> <condition field="${proto_specific_hangup_cause}" expression="<a href="sip:408">sip:408"> <action application="set" data="sip_ignore_remote_cause=true"/> <action application="respond" data="503"/> <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/> </condition> </extension> How to correctly remap causes? The wiki talk about the fact that is not possible to remap causes after a 180 or 183, is this true? If there is no way to change normal behaviour, do you think could be possible also to change the source code and recompile FS accordly? Thank yo so much and sorry for this long post I hope someone could help Max

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Codec Error!
by Ali R. 03 Sep '14

03 Sep '14

Hello All, I'm not sure if this must be addressed to the user list but I apologize in advance. I'm getting the below error while in pass-through mode (proxy_media=true) but I get success when the disable transcoding is set on the sofia profile. [WARNING] switch_core_codec.c:703 Codec G723 Exists but not at the desired implementation. 8000hz 20ms [ERR] switch_core_media.c:2285 Can't load codec? I did hear about broken SIP devices out there but since it works fine when disable transcoding is set, I thought I give it a shot on the mailing list and while at it, this error does not occur in case of another pass-through codec(G729) Here are my settings when in proxy media mode: Sofia Profile: inbound-late-negotiation = true inbound-codec-negotiation = greedy (tried scrooge) I also tried rtp-autofix-timing=true (no luck) DialPlan: proxy_media=true bridge FS version tried: 1.5.13b & 1.2 Attached is full FS log for both the failed & successful calls. Your help or suggestion is very much appreciated. Best Regards, Ali_R

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Please do not commit directly to v1.4 branch at this time
by Ken Rice 03 Sep '14

03 Sep '14

We are updating the v1.4 branch incrementally. Please do not commit or cherry pick patches over to this branch at this time K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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FreeSWITCH to Implement Video Transcoding - Sponsors Needed
by Anthony Minessale 02 Sep '14

02 Sep '14

The FreeSWITCH project intends to devote attention in the upcoming months to work towards improving video functionality. This includes working to get fully function video transcoding working inside FreeSWITCH. Currently FreeSWITCH video is limited to pass-through and with the new features available with WebRTC we have decided to focus on trying to improve the overall video capabilities in FreeSWITCH. We currently have an issue open with chrome that helps underscore the reason we need to implement native video codec functionality especially in Conferencing. https://code.google.com/p/chromium/issues/detail?id=263867 We anticipate that this effort will take as long as 6 or more months to complete and have a large cost in developer resources. We already have some sponsors and volunteers helping with the effort but we can use all the help we can get. If you are interested in helping out either by financially sponsoring the effort or helping to test and bug report, please contact us at consulting(a)freeswitch.org and let us know. -- 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

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Implementing preferred channel selection in freetdm using sangoma isdn ?
by Matteo 01 Sep '14

01 Sep '14

Hi, the following applies to ftmod_sangoma_isdn. Scenario: some pri isdn switches does not like exclusive channel selection, but needs the setup message to carry a preferred channel selection ie. Adding this in freetdm (in setup message) and making it configurable is easy and an example patch is attached. BUT! If the remote switch changes the channel, we need to "move" the current freetdm channel to the one specified in the provisional response or setup ack. According to q.931 is wrong to allocate a b channel before having confirmed the right one to use, but right now freetdm allocates it immediately. (maybe is worth changing that?) So, I'm trying to understand all the stuff between freetdm and ftmod_sangoma_isdn in order to modify it and make the preferred b channel selection working. Any hint ? Which is the best way to "swap" a freetdm channel or move to another? Is something the stack already allows or needs some major modifications? mat

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FreeSWITCH Friday BugHunt Reminder!
by Ken Rice 29 Aug '14

29 Aug '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken http://ift.tt/vGSsGm http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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Allowing calls with Mod nibblebill and zero balances
by Roland Selmer 28 Aug '14

28 Aug '14

I’ve got nibblebill working quite well and it’s a great little mod. However one use case I can’t get to work is allowing calls to happen when a users balance is 0 and I’d like to allow calls if they are ‘internal’ i.e. not using a gateway. Although I set nibble_rate=0, the dial plan doesn’t even get there as it sees a 0 balance and hits the hangup extension. Has anybody this use case working?

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