May 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

sending custom verto events to client
by Vik Killa 12 May '15

12 May '15

Hello, We have verto setup and working. It is great btw. We want to be able to send custom json data back to the client (hopefully using dialplan if possble) I don't really know if this is even possible or if it is, where to start. I change mod_verto to verbose debugging and I can see it sends json data back to the verto client, but I am lost as to how to do this on my own. Here is an example json sent from FS to verto client: 2015-05-12 05:18:04.470641 [ALERT] mod_verto.c:604 WRITE 192.168.10.101:48165 [{ "jsonrpc": "2.0", "id": 8, "result": { "message": "CALL CREATED", "callID": "be3e35a0-ec02-5ca7-cbf8-68a8c1cd2c4c", "sessid": "48a502b3-740a-5b3d-21a4-bf9244bafb90" } }] I see there is a "json" API command in FS console but when i try to run something like json {"message":"CALL CREATED","callID":"be3e35a0-ec02-5ca7-cbf8-68a8c1cd2c4c","sessid":"48a502b3-740a-5b3d-21a4-bf9244bafb90"} I get {"message":"CALL CREATED","callID":"be3e35a0-ec02-5ca7-cbf8-68a8c1cd2c4c","sessid":"48a502b3-740a-5b3d-21a4-bf9244bafb90","status":"error","message":"Invalid request or non-existant command","response":null} If someone know how I can accomplish this, I'd be happy to update confluence with details. Thanks, V

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mod_callcenter: wrap_up_time
by Karolis Pabijanskas 12 May '15

12 May '15

Hi All, I am using a mod_callcenter for this particular set-up. I need the agents to have a wrap_up_time set to, say, 300 seconds, which works great. In some cases, however, the agent needs to have a code he can dial so he resets his wrap_up_time (say, he receives a call to the wrong number). It does seem that if I call: callcenter_config agent set state AGENT Waiting It sets the call center to available and the wrap-up time is cancelled, but the wrap-up never kicks in again! Is there a way to reset the agent wrap_up_time via the API (which I can then call via the dialplan) so that the agent is available for a new call immediately, but that the wrap-up time kicks in afterwards as normal? Many thanks for your help! Karolis

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FreeSWITCH Week in Review (Master Branch) May 2nd-May 8th
by Ken Rice 11 May '15

11 May '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/1bKCY9S FreeSWITCH Week in Review (Master Branch) May 2nd-May 8th Hello, again. This passed week in the FreeSWITCH master branch we had 6 commits. Some more work was done to mod_amqp this week as well as some bug fixes. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-7526 Add enable_fallback_format_fields for mod_amqp producer profiles if the profile param is set and create the amqp exchange on the first startup of a clean platform. The following bugs were squashed: FS-7425 Fixed a bug when using a cert with missing dhparams resulting in a segfault. FS-7523 [mod_json_cdr] Fixed a segfault caused by a missing config file. FS-7357 FAX now tolerates EOP and PPS messages being incorrectly echoed.

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need SIP IP / RTP recommendation
by Jason Holden 11 May '15

11 May '15

Currently I have a client who is using multiple WAN IP addresses. On route fail over FS still tries to use the original IP details from the external profile and not the secondary circuit details from the secondary external profile. What have others done to address this to ensure failover works propperly?

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bind_digit_action for lua
by Aqs Younas 11 May '15

11 May '15

Hi, users. Below is my default.xml snippet. I am using bind_digit_action for different inputs bindings and moves the call to that context upon regex match. <condition> <action application="bind_digit_action" data="moderator,~\*,exec:execute_extension,previous XML Previous"/> <action application="bind_digit_action" data="moderator,~\#,exec:execute_extension,next XML Next"/> <action application="bind_digit_action" data="moderator,~^\d$,exec:execute_extension,exten XML Exten"/> <action application="digit_action_set_realm" data="moderator"/> <action application="answer"/> <action application="playback" data="shout://127.0.0.1/radio"/> </condition> Now I want to write my dialplan in lua which I think is more handy than xml. Upon regex match call to be executed in corresponding functions. Like, for * *function next()* for #, *function previous()* I know there is a application *session.setHangupHook(hangup_function_name)*; Which executes the function(hangup_function_name) upon call hangup. Is there anything available in lua witch executes functions upon regex match. Or any other way for doing this. Any pointer would be much appreciated. Thanks.

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info about sofia sip dns cache and graylisting?
by Michel Brabants 11 May '15

11 May '15

Hello all, we have a problem with a 1.4-version of Freeswitch that FS goes to 100% cpu. The problem has been traced to a dns-cache-problem. The dns-cache contains srv-records, which should have 2 a-records, but they have 100's of a-records, which are mostly the same (some with a different priority). This causes FS do to a lot of compares (to sort the records) and it goes to 100% cpu in the end. The priority almost never is the original priority, so I was thinking that the cache is maybe growing because of graylisting of entries, causing the priority to increase and the ttl to be reset. The code is however very difficult to follow as it is all event-based and part of big C-structs in sofia-sip. A bug in the gdb-debugger causing static variables to be almost unreadable makes life not much easier. The graylisting policy, as well as the dns cache policy is not really documented as far as I can find, so any info about this would be great. Anyway, my current thought is that the dns-cache is maybe growing, becuase the graylisting-code sets the priority to a higher value and resets the ttl to a certain value (no idea which one, but the default is 10 minutes and the maximum a day). The srv-record-compare-function however checks also using priority and weight and returns a negative response if they are different. I would think that fs maybe then readds the original srv-records (with priority 0) when it requeries the dns. Those entries will also increase in priority and ttl when they are added, causing the cache to keep growing ... Nscd is also being used for dns-caching, which doesn't help, but I'm not sure if that is the problem as sofia_dig seems to return valid values (except maybe ttl). I would think that ttl is never touched and that the cached is cleared when the ttl expires, but I have no idea when the dns-cache is cleared in FS... I know this is maybe a lot of text, but any info on this topic (sofia dns cache) is welcome. Thanks, Michel

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Re: [Freeswitch-users] FEC feature in mod_opus
by Kamil Nigmatullin 11 May '15

11 May '15

Hi Dragos, I tried to use your patch but have no luck with compiling it. I clonne stable 1.2 and then replace mod_opus.c with your code. Maybe do you know how to fix it? This is what I get: d_opus/mod_opus.c -fPIC -DPIC -o .libs/mod_opus.o /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c: In function 'switch_opus_fmtp_parse': /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c:155:18: error: 'switch_codec_fmtp_t' has no member named 'stereo' /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c: In function 'switch_opus_decode': /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c:423:3: error: statement with no effect [-Werror=unused-value] cc1: all warnings being treated as errors make[4]: *** [mod_opus.lo] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_opus-all] Error 1 make[1]: *** [mod_opus] Error 2 make: *** [mod_opus] Error 2 2015-04-14 14:52 GMT+06:00 Dragos Oancea <droancea(a)yahoo.com>: > > Hi Kamil, > > I've done a patch for Opus with FEC , but it has not been merged: > > https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/184… > I was inspired by how they've done it with SILK. > > > Regards, > Dragos Oancea > > > > On 2015-04-13 16:49, Kamil Nigmatullin wrote: > > Hello. > > > > Anybody from developers could please clarify if FreeSWITCH uses > > Forward Error Correction feature in opus codec? We actually are not a > > specialists in C and cannot fully understand how decode function works > > in 414 line. According to opus documentation decode function is > > called with special flag if previous packet is lost in order to call > > fec feature. In mod_opus it is hard to say for me as i don't see i or > > i+1 packets. Thanks > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin

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Trickle ICE with SIP
by Adam Ben-Ayoun 11 May '15

11 May '15

Hi, We are currently using Freeswitch for voice conferencing, we use SIP for signalling. Call setup times are really slow at times (3sec-30sec), can we somehow do Trickle ICE with Freeswitch? Thanks, Adam

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Clock precision and raw audio recording
by Stanislav Sinyagin 10 May '15

10 May '15

I'm observing an effect which needs explanation. Comments from the core developers will be appreciated. The effect was tested with versions 1.4.18 and today's master, on 64-bit Debian 7 and Ubuntu 14.04. All test calls were in PCMU or PCMA. My customer requested me to build a server for call quality assurance for their telephony system. I installed FreeSWITCH and set the following in the public dialplan to record the incoming audio: <extension name="record"> <condition field="destination_number" expression="^record_(.+)$"> <action application="jitterbuffer" data="60:200:20"/> <action application="set" data="RECORD_READ_ONLY=true"/> <action application="set" data="send_silence_when_idle=400"/> <action application="set" data="record_waste_resources=true"/> <action application="answer"/> <action application="record_session" data="/var/tmp/record_$1"/> <action application="playback" data="silence_stream://-1"/> </condition> </extension> The first try was with a DigitalOcean (KVM) virtual machine. I started recording *.wav files, and sometimes there were skipped frames: 1-2 skipped frames in a 2-minute call, one in every 10-15 calls. Then I changed the configuration to write raw audio files (removed the .wav extension from the record_session argument). As a result, the recorded input audio was quite bad: lost frames every few seconds in every call. Then I made test calls within the server itself, by originating a call to its public profile and playing the test WAV audio: fs_cli -x 'originate sofia/external/record_03@111.222.222.111:5080 &playback(/var/tmp/ITU-T_P_50_BRITISH_ENGLISH.wav)' The resulting input raw audio was also choppy. The same result was on another VM on a different physical server at DigitalOcean. Then I made the same self-call tests on a Xen VM and on a baremetal ARM server, and there the recorded audio was of perfect quality. Self-calls with recording into WAV files produced audio of perfect quality. Setting RECORD_USE_THREAD=false did not change the effect. Example of choppy received audio, converted from PCMU to WAV for convenience: http://www.k-open.com/s/record_04-in.wav The source audio: http://murmur.voxserv.ch/media/ITU-T_P_50_BRITISH_ENGLISH.wav So, it looks like the clock that is available at KVM is not precise enough, but that's not my question. QUESTION: why is raw recording so much more sensitive to the clock precision? thanks, stanislav

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Connect two freeswitch boxes with gateway
by Luiz Fernando Softov 10 May '15

10 May '15

Hi, i'm trying to do this B have a gateway registered with user 1001 in A and of course, A have a user 1001 registered B ----> [gateway] ------> A (1001) So, i'm trying to bridge 1001 when i call (1xxx) A ----> [bridge user/1001] ------> B and have this dial-string for user 1001 {presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})} In FS B i receive the call, but i want to receive the real destination number, like 1002 or 1004 Is there a way to bridge to user, and send the destination number? So, in box B, i can transfer to correct user. -- Luiz Fernando Softov http://www.softov.com.br fernando(a)softov.com.br

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