June 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Now that 1.4.19 and the fs-video2 branch merged to master...
by Brian West 07 Jun '15

07 Jun '15

We cordially invite the community: [image: Inline image 1] And remember, If you think its a bug, file a JIRA, if its docs we'll move those to the docs project to be corrected or writte. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> | Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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Language Module / Default config Missing say-module
by Victor Medina 07 Jun '15

07 Jun '15

Hi guys! First of all hope you are having a great weekend! I noted while working with the language module, that the default config is missing the "say-module" tag. The languages missing the the say-module config are (that I tested) DE ES PT_BR PT_PT RU SV I can fill jira if you want. Just let me know. -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys

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Advanced channel mix
by Dmitry Mordovin 06 Jun '15

06 Jun '15

Hello, Need advanced "channel bridge/mix" solution. Scenario: - user 1000 called to 1005 and talks - user 1005 hold 1000 and make second call to 1007 At this point we have three users and two active calls (sessions) first call: users 1000 (a leg) and 1005 (b leg) second call: user 1005 (a leg) and 1007 (b leg) - mix users. user 1000 (first call, a leg) connect to user 1007 (second call, b leg) - send BUY to user 1005 for two calls - user 1000 and 1007 continue talks I thing this "mix" command may be like "eavesdrop" but with two uuids of channels or else Anybody try to implement same scenario?

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Setting REGISTER expiration time
by Juan Antonio Ibañez Santorum 05 Jun '15

05 Jun '15

Hello, I can see my FS box is asnwering to REGISTER messages with a 30 secs expiration time: Contact: <sip:736484354451@7.15.6.20:5061>;expires=30 Which would be the rigth way to increase that value? I can see variable "sip-force-expires" but I don't know if it is the right one Regards

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Setting loglevel beyond 7 ?
by Sharath Kumar 05 Jun '15

05 Jun '15

All, I noticed some of the logs in the code are using SWITCH_LOG_DEBUG1 =101 - SWITCH_LOG_DEBUG10 = 110. Is there a way to enable these logs from the fs_cli ? I can recompile the ones I want with a lower value but just wanted to avoid that. Thanks Sharath

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High Availability for Webrtc/DTLS ?
by Sharath Kumar 05 Jun '15

05 Jun '15

Hi, I am using Freeswitch for terminating DTLS traffic while the webrtc/websocket is terminated by a Kamailio front-ending the FS. Assuming I have HA configured as given here https://freeswitch.org/confluence/display/FREESWITCH/High+Availability Can I successfully recover calls on the secondary using 'sofia recover' when the primary crashes ? My gut feel is it won't work since the DTLS context probably needs to be sent to the secondary as well ? Perhaps the secondary needs to do a new offer-anwer and regneg DTLS ? Thoughts? I am going to try this anyway but if someone knows it will save me a ton of time. Thanks, Sharath

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rfc2833 DTMF behaviour
by Pete Kelly 05 Jun '15

05 Jun '15

Hi I am trying to send rfc2833 DTMF with SIPP into FreeSWITCH and have the DTMF be passed along to the bridged leg. If I set up the following scenario: Bria > FreeSWITCH > SIPP and press a DTMF tone on Bria, I see the DTMF received in FS and then it being sent out to the bridged leg. However if I replace Bria with SIPP and play one of the SIPP supplied rfc_2833 pcaps, I only see the DTMF being received by FreeSWITCH but it is not sent to the bridged leg. Does anyone have any suggestions? This is the log indicating FS is receiving the DTMF from SIPP 2015-06-04 15:41:57.343854 [DEBUG] switch_rtp.c:6092 RTP RECV DTMF 2:2240 2015-06-04 15:41:57.343854 [DEBUG] switch_channel.c:488 RECV DTMF 2:2240

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DTMF INFO not configuring
by Dmitry Mordovin 05 Jun '15

05 Jun '15

Hello, Try to configure DTMF INFO - don't works.... FS default configuration, User 1000 calling to sofia gateway (SIP provider) I set in external.xml and internal.xml <param name="dtmf-type" value="info"/> Set in dialplan <action application="set" data="dtmf_type=info"/> Restart FS, still not works. Console: 2015-06-04 11:54:22.977598 [WARNING] sofia.c:8617 IGNORE INFO DTMF(1) (This channel was not configured to use INFO DTMF!) My be I forget something in configuration?

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 05 Jun '15

05 Jun '15

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 FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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I've broken my moh
by rick.jarvis＠magicmail.mooo.com 04 Jun '15

04 Jun '15

Hi all, I’m getting: EXECUTE sofia/internal/501(a)10.0.0.1 playback(local_stream://moh) 2015-06-01 22:35:59.233311 [WARNING] mod_local_stream.c:470 Unknown source moh, trying 'default' 2015-06-01 22:35:59.233311 [ERR] mod_local_stream.c:479 Unknown source default The wav files are there, mod_local_stream is loaded, and local_stream.conf.xml is the same as vanilla. Must be missing some config part, I’ve even grep’d vanilla for ‘moh’ to see if I can find something I haven’t got, but that hasn’t helped. What else can I check? R

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