June 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

XML curl "not found" response
by Vladyslav Zakhozhai 08 Jun '17

08 Jun '17

Hi, I am curious what HTTP status xml curl server must provide when no result is found for request with body <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="result"> <result status="not found" /> </section> </document> I mean HTTP 200 or HTTP 404. My problem is the following: 1. When freeswitch gets 404 for directory request it reports error: [ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch 2. When freeswitch gets 200 with status "not found" on dialplan request failover to file xml config does not occure. Thank you in advance. -- С уважением, Владислав Захожай

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BugHunt
by Sergey Safarov 08 Jun '17

08 Jun '17

Is BugHunt will be today?

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Re: [Freeswitch-users] Turn configuration mod_sofia
by Alexander Haugg 08 Jun '17

08 Jun '17

Hi, my question: how can I configure an ICE server (turn) in the sip profile? My think for this id case I need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too. - STUN configuration "<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org"/>" -> that's working! - ICE configuration could be? "<param name="ext-rtp-ip" value=" { 'stun:stun.freeswitch.org', 'turn:turn.myserver.com;username=xxx;password=123'}"/>" Is there a possibility? Thanks a lot Von: Alexander Haugg Gesendet: Donnerstag, 1. Juni 2017 15:24 An: 'freeswitch-users(a)lists.freeswitch.org' <freeswitch-users(a)lists.freeswitch.org> Betreff: WG: Turn configuration mod_sofia Hi Brian, we use the Freeswitch as "Man in the Middle" for WebRTC. The WebRTC Clients (ICE Link Stack from Frozen Mountain) registered via SIP on the Freeswitch. The SIP signalling is a special szenario and works successfully in several LAN WAN setups. But now we need srflx (that's fine with teh STUN configuration) an relay candidates in the SDP that's ganerated by the Freeswitch. My think is, for this case i need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too. - STUN configuration "<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org"/>" - ICE configuration could be? "<param name="ext-rtp-ip" value=" { 'stun:stun.freeswitch.org', 'turn:turn.myserver.com;username=xxx;password=123'}"/>" Thanks a lot >From brian at freeswitch.org Tue May 30 13:50:32 2017 From: brian at freeswitch.org (Brian West) Date: Tue, 30 May 2017 08:50:32 -0500 Subject: [Freeswitch-users] Turn configuration mod_sofia In-Reply-To: <a9e067a18e104afbb53e207712467baf(a)c4b.de<mailto:a9e067a18e104afbb53e207712467baf@c4b.de>> References: <a9e067a18e104afbb53e207712467baf(a)c4b.de<mailto:a9e067a18e104afbb53e207712467baf@c4b.de>> Message-ID: <CAEJMVkDy5e2COC8cH=xV=ozM0gepBS3cQ_YxUPQk4twMoaSVmg(a)mail.gmail.com<mailto:CAEJMVkDy5e2COC8cH=xV=ozM0gepBS3cQ_YxUPQk4twMoaSVmg@mail.gmail.com>> There is little reason to use TURN when speaking to FreeSWITCH, What issue are you trying to solve? Von: Alexander Haugg Gesendet: Montag, 29. Mai 2017 18:25 An: 'freeswitch-users(a)lists.freeswitch.org' <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Betreff: Turn configuration mod_sofia Hi all, the stun configuration for the mod_sofia profile is very easy "<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org"/>" but what is to do, if i need a relay candidate in the sdp? How can i set the turn address and the login credentials? Thanks a lot

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Re: [Freeswitch-users] Enabling new languages for Say
by boesemar . 07 Jun '17

07 Jun '17

Hi Brian, mod_say_pt is definitely loaded: > module_exists mod_say_pt true Can you think of any other cause for the "Invalid SAY Interface [pt]" ? Martin PS: sorry for breaking the thread - problem with email Brian West brian at freeswitch.org Fri Jun 2 14:51:25 UTC 2017 I would guess mod_say_pt is not loaded. On Fri, Jun 2, 2017 at 4:54 AM, Martin Boese <martin at maxnet.ao <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote: >* Hi! *>>* FreeSWITCH Version 1.6.15-32-bec4538~64bit (-32-bec4538 64bit) *>* Debian Jessie. Vanilla config. *>>* I am trying to enable portuguese to say numbers using dptools "Say". *>>* This is what I did: *>* - Module mod_say_pt is loaded *>* - in freeswitch.xml *>* include lang/pt/pt_PT.xml *>* - Downloaded sounds from *>* https://github.com/jpawlowski/freeswitch-sounds-ttsand <https://github.com/jpawlowski/freeswitch-sounds-ttsand> placed them *>* into /usr/share/freeswitch/sounds *>* Directory structure is now like: /pt/tts/google/ascii/16000/... *>>* CLI> say_string t.wav pt NUMBER pronounced 123 *>* [ERR] switch_xml.c:3180 Can't find phrases tag *>>* ..I found out that lang/pt_PT.xml seems to be missing the *>* <phrases><macros> tags within the <language> tag (vanilla config). I *>* fixed that. *>>* Now: *>* CLI> say_string t.wav pt NUMBER pronounced 123 *>* [ERR] switch_ivr.c:3726 Invalid SAY Interface [pt]! *>>* BTW: English works fine: *>* CLI> say_string t.wav en NUMBER pronounced 123 *>* file_string://digits/1.wav!digits/hundred.wav!digits/20.wav!digits/3.wav *>>* I also tried other languages but have same error "Invalid SAY *>* Interface". *>>* What am I missing. Please help. *>>* Thanks, *>* Martin *>>* _________________________________________________________________________ *>* Professional FreeSWITCH Consulting Services: *>* consulting at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> *>* http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com> *>>* Official FreeSWITCH Sites *>* http://www.freeswitch.org <http://www.freeswitch.org> *>* http://confluence.freeswitch.org <http://confluence.freeswitch.org> *>* http://www.cluecon.com <http://www.cluecon.com> *>>* FreeSWITCH-users mailing list *>* FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> *>* http://www.freeswitch.org <http://www.freeswitch.org> * -- *Brian West*brian at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.comhttp://www.freeswitchcookbook.com Book a phone call (CST) <https://freeswitch.com/appointment> Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest

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Monitoring call quality using variable_rtp_audio_xxx
by Jonathan Hunter 07 Jun '17

07 Jun '17

Hi Guys, Sorry for the noise, we are looking to poll calls in progress to grab the rtp_audio variables when a call is in progress, and we want to understand if this is a good approach and to that end what values/variables we should consider and what ranges should we be working with? I understand mos and quality percentage but what other values are a good indicator? Again Im clear on jitter and its meaning just want to understand whats what as not everything understandably is not documented. Many thanks Jon "variable_rtp_audio_recv_pt": "8", "variable_rtp_audio_in_raw_bytes": "374960", "variable_rtp_audio_in_media_bytes": "374616", "variable_rtp_audio_in_packet_count": "2180", "variable_rtp_audio_in_media_packet_count": "2178", "variable_rtp_audio_in_skip_packet_count": "92", "variable_rtp_audio_in_jitter_packet_count": "0", "variable_rtp_audio_in_dtmf_packet_count": "0", "variable_rtp_audio_in_cng_packet_count": "0", "variable_rtp_audio_in_flush_packet_count": "2", "variable_rtp_audio_in_largest_jb_size": "0", "variable_rtp_audio_in_jitter_min_variance": "28.57", "variable_rtp_audio_in_jitter_max_variance": "116.33", "variable_rtp_audio_in_jitter_loss_rate": "0.02", "variable_rtp_audio_in_jitter_burst_rate": "0.98", "variable_rtp_audio_in_mean_interval": "20.39", "variable_rtp_audio_in_flaw_total": "43", "variable_rtp_audio_in_quality_percentage": "97.00", "variable_rtp_audio_in_mos": "4.47", "variable_rtp_audio_out_raw_bytes": "210012", "variable_rtp_audio_out_media_bytes": "210012", "variable_rtp_audio_out_packet_count": "1221", "variable_rtp_audio_out_media_packet_count": "1221", "variable_rtp_audio_out_skip_packet_count": "0", "variable_rtp_audio_out_dtmf_packet_count": "0", "variable_rtp_audio_out_cng_packet_count": "0", "variable_rtp_audio_rtcp_packet_count": "0", "variable_rtp_audio_rtcp_octet_count": "0",

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trouble with instalation of mod_fail2ban module in FreeSwitch.
by Miguel Jesús López Valverde 06 Jun '17

06 Jun '17

Hello: I am sending you this email because I am having problems installing the module mod_fail2ban following the recipe offered at https://freeswitch.org/confluence/display/FREESWITCH/mod_fail2ban because when I run make, I get this result: /usr/local/freeswitch/mod/applications/mod_fail2ban$ sudo make Makefile:2: ../../../../build/modmake.rules: No such file or directory make: *** No rule to make target '../../../../build/modmake.rules'. Stop. Do you know in which folder the sentence git clone may to be executed?. Do you know if there is anything else to keep in mind that can cause this problem? Thank you very much. Miguel J. Lopez. --- El software de antivirus Avast ha analizado este correo electrónico en busca de virus. https://www.avast.com/antivirus

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X-RTP-STAT or P-RTP-STAT
by William Simon 06 Jun '17

06 Jun '17

Can anyone explain how to add the X/P-RTP-STAT header to the BYE for summarizing RTP statistics at the end of the call? Seems like there's something built in for this, but it's not clear how to use it. https://wiki.freeswitch.org/wiki/Variable_sip_p_rtp_stat "The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer."

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Fwd: How to get if of the playing media in a conference
by Deepika Yadav 06 Jun '17

06 Jun '17

---------- Forwarded message ---------- From: "Deepika Yadav" <deepikay(a)iiitd.ac.in> Date: Jun 6, 2017 4:29 PM Subject: How to get if of the playing media in a conference To: <freeswitch-users-owner(a)lists.freeswitch.org> Cc: Hi, I need to detect the start and stop of media in a conference call through ESL. Event named as "play-file-done" and "play-file" are for detection of any kind of media related activity. However, I could get find a method to get the ID of the media to identify them. e.g. event "play-file-done" is detected multiple times in a conference - if someone is added and also when any playing media is stopped. -- Regards Deepika https://www.iiitd.edu.in/~deepikay/

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About proxy-hold codec re-negotiation behavior.
by Lợi Đặng 06 Jun '17

06 Jun '17

Hi list, I'm using FS with *proxy-hold*. Call flow is simple A -> FreeSWITCH -> B Assume FS, A and B support PCMA and PCMU, FS prefer PCMA in codec-prefs profile configuration. In affects of my configuration for *greedy early negotiation* and *disable transcoding*, PCMA is negotiated in both legs for initial INVITE, that's good, and expected. But then I remove PCMA from A and compose a re-INVITE for holding with PCMU only, the a leg is re-negotiated with PCMU, but the b leg have re-INVITE with PCMA due to *proxy-hold* variable, that causes transcoding happened after hold. My desire is to have FS to re-negotiate with b leg too, not so similar but as what I observed when I was using it with media proxy option ( A-A before hold, U-U after hold). I tried late negotiation in this case, but no luck. So is that not able for FS to re-negotiate codec on b leg, in using proxy-hold? Any advise is appreciated. rgds, Loi Dang

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Videoconference error with H263+ [ERR] avcodec.c:736
by Andres Gomez 05 Jun '17

05 Jun '17

Hello. I see this console error videcoconference with H263+ 2017-06-05 08:14:44.668608 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 d2 ef 2017-06-05 08:14:44.668608 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 f4 12 2017-06-05 08:14:44.668608 [ERR] avcodec.c:736 len: 382, mark:1 00 00 01 c0 2017-06-05 08:14:44.678607 [ERR] avcodec.c:736 len: 1198, mark:0 04 00 82 ca 2017-06-05 08:14:44.678607 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 56 83 2017-06-05 08:14:44.678607 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 24 4e 2017-06-05 08:14:44.678607 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 e9 db 2017-06-05 08:14:44.678607 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 9a 3a Any Idea why? Regards

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