January 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

MOH from live URL
by Uday kumar 25 Jan '18

25 Jan '18

Hi, Is possible to play MOH from live URL or any radio station? I have few URL that I can directly on browser. In Asterisk I used to do with Playback like /usr/bin/mpg123 -q -r 8000 -f 8192 --mono -s https://URL So though it may be possible in Freeswitch. Please guide. -- Thanks & Regard Uday. Mobile:- +91-9377579349

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fax call fails with repeated log message blows up logs
by Douglas Davenport 25 Jan '18

25 Jan '18

randomly we get this message over and over on outbound fax calls. Any ideas what causes fax to end up in this state? Running FS 1.6.18 2018-01-24 15:49:22.368959 [DEBUG] mod_spandsp_fax.c:276 FLOW T.30 Start T4 2018-01-24 15:49:22.368959 [DEBUG] mod_spandsp_fax.c:276 FLOW T.38T No new frame or end transmission condition. 2018-01-24 15:49:22.368959 [DEBUG] mod_spandsp_fax.c:276 FLOW T.30 Send complete in phase D_RX, state II_Q 2018-01-24 15:49:22.368959 [DEBUG] mod_spandsp_fax.c:276 FLOW T.30 Changing from phase D_RX to D_RX 2018-01-24 15:49:22.368959 [DEBUG] mod_spandsp_fax.c:276 FLOW T.38T Set rx type 4 2018-01-24 15:49:22.368959 [DEBUG] mod_spandsp_fax.c:276 FLOW T.38T Set tx type 0 2018-01-24 15:49:22.368959 [DEBUG] mod_spandsp_fax.c:276 FLOW T.30 Start T4 2018-01-24 15:49:22.368959 [DEBUG] mod_spandsp_fax.c:276 FLOW T.38T No new frame or end transmission condition. 2018-01-24 15:49:22.368959 [DEBUG] mod_spandsp_fax.c:276 FLOW T.30 Send complete in phase D_RX, state II_Q 2018-01-24 15:49:22.368959 [DEBUG] mod_spandsp_fax.c:276 FLOW T.30 Changing from phase D_RX to D_RX seems to relate to this line in source: https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/libs/spand…

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Re: [Freeswitch-users] spandsp_start_tone_detect is compatible with G729 codec???
by Raimundo Pérez Nieves 24 Jan '18

24 Jan '18

Thanks Brian, but I am using tone_detect with g729 (commercial license) and works very good. That was the reason I tried with spandsp module, because it allows detect tone using cadence. What is the meaning of f1 and f2 in segment tone debug, nobody can answer me this question. Thanks Enviado desde mi iPhone El 24/01/2018, a la(s) 16:40, Brian West <brian(a)freeswitch.com> escribió: > You can't do tone detect on G729, the bandwidth of the audio is insufficient! > >> On Wed, Jan 24, 2018 at 5:58 AM, Raimundo Pérez Nieves <raimundo.perez.cuba(a)gmail.com> wrote: >> Hello, >> spandsp_start_tone_detect is compatible with G729 codec??? >> Because Iam testing and I get no result. >> >> 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 > > > > -- > > Brian West | Co-founder and Developer > Need Commercial support? email sales(a)freeswitch.com > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > Email: brian(a)freeswitch.com > Mobile: 918-424-9378 > Website: https://www.FreeSWITCH.com

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[ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error]
by Raimundo Pérez Nieves 24 Jan '18

24 Jan '18

Hi guys, I put shutdown in freeswitch but did't change nothing and when I fs_cli I get this error [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error]

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spandsp_start_tone_detect is compatible with G729 codec???
by Raimundo Pérez Nieves 24 Jan '18

24 Jan '18

Hello, spandsp_start_tone_detect is compatible with G729 codec??? Because Iam testing and I get no result. Thanks

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freeswitch_licence_server change location where to scan licences
by António Silva 24 Jan '18

24 Jan '18

Hi, Is it possible to change the default location "/etc/freeswitch" where the freeswitch_licence_server is scanning? It would make it faster to start, because now it scans the entire dir and the default configuration is now stored in this dir. In logs i see: Jan 24 14:09:30 main freeswitch_licence_server[14081]: Scanning /etc/freeswitch/directory/default/1000.xml for licences Jan 24 14:09:30 main freeswitch_licence_server[14081]: Scanning /etc/freeswitch/directory/default/1012.xml for licences Jan 24 14:09:30 main freeswitch_licence_server[14081]: Scanning /etc/freeswitch/directory/default/1015.xml for licences Jan 24 14:09:31 main freeswitch_licence_server[14081]: Scanning /etc/freeswitch/directory/default/1005.xml for licences Jan 24 14:09:31 main freeswitch_licence_server[14081]: Scanning /etc/freeswitch/directory/default/1008.xml for licences -- Saludos / Regards / Cumprimentos António Silva

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Clarification regarding register behavior
by Kai Danielmeier 24 Jan '18

24 Jan '18

Hi all, I currently facing a problem with frquent changes in SRV record weight and a freeswitch behavior, which causes some trouble in this regard. Provider is Deutsche Telekom and for registration the SRV record reg.sip-trunk.telekom.de is used. Due to load balancing this records change frequently during the day. When a change occurs freeswitch send the INVITE requests to the "new" ip gathered from DNS and gets a 403 denied message back from server host. Analyzing the SIP traces I found that freeswitch is only doin DNS requery for INVITE, OPTIONS ping etc., but not for REGISTER messages. REGISTER is always going to the host gathered from the SRV record during gateway start even after expiry. (My first try was to set expiry to minimum allowed by server host, which is 240 seconds and hoping for FS to requery DNS and then register with new server host, but this did not work due to mentioned reasons) Is this the correct behavior or is this a bug? Is there an option switch somewhere to influence this? Thanks for a feedback. Kind regards Kai Danielmeier

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Tone segment f1 and f2
by Raimundo Pérez Nieves 23 Jan '18

23 Jan '18

Hi guys What is the meaning of this (specially f1 and f2): mod_spandsp_dsp.c:705 Tone segment: f1 = -1, f2 = -1, duration = 11232 mod_spandsp_dsp.c:705 Tone segment: f1 = 1, f2 = 3, duration = 48 mod_spandsp_dsp.c:705 Tone segment: f1 = -1, f2 = -1, duration = 944 mod_spandsp_dsp.c:705 Tone segment: f1 = 2, f2 = 3, duration = 32 mod_spandsp_dsp.c:705 Tone segment: f1 = 3, f2 = -1, duration = 32 mod_spandsp_dsp.c:705 Tone segment: f1 = -1, f2 = -1, duration = 944 mod_spandsp_dsp.c:705 Tone segment: f1 = 1, f2 = 2, duration = 64

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start_dtmf_generate not working with opus
by Varghese Paul 23 Jan '18

23 Jan '18

Hi all, I am facing a issue with start_dtmf_generate when i use opus codec and SIP INFO dtmf. Caller ( opus/dtmf ) ------------------------ Freeswitch ----------------------------- Calleee ( ulaw/inband) Caller is using jssip webrtc and dtmf is in INFO. For inband dtmf generation i am using start_dtmf_generate in caller channel. The setup is working when i use ulaw or ilbc or alaw codec. When i use webrtc with OPUS codec i am seeing the inband dtmf generation is not working. Webrtc with ulaw codec and dtmf INFO is working fine. Version : FreeSWITCH Version 1.6.17~64bit ( 64bit) I am attaching the opus.conf.xml for the reference. Any idea, why start_dtmf_generate is not working with opus codec ?. Do i need to set any setting for opus ?. Regards Varghese Paul

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SIP 482 "Request Merged" on parallel forking
by Carlos Ruiz Díaz 23 Jan '18

23 Jan '18

Hey guys, It seems like FS is unable to process a parallel forking originating in Kamailio (or others proxies of its type). Kamailio does the routing logic, and creates various branches all with different request URIs but keeping the same CSeq and Call-ID and changing the branch parameters, but FS interprets these requests as the same and merges them running the forking impossible. - I've seen replies that the CSeq should be increased, but this is not correct according to the RFC. - Change Call-IDs, which should not be the case for parallel forking. - And the only reported solution, change the T4 SIP timer to less than 4 seconds and put a sleep() on Kamailio for each branch. I personally don't consider this a solution as it plays with the timers. Do you have any recommendations or ideas to solve this? Thanks, -- Carlos

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