- freeswitch-dev@lists.freeswitch.org - FreeSWITCH Mailing Lists

TTS Usage with Cepstral Voices
by Maverick Good 09 Oct '19

09 Oct '19

Hello, I would like to add TTS to my Freeswitch install. I bought two voices from Cepstral previously to use with this. The way it should work is a user enters a TTS string in a web page field and when the page is submitted we create the audio file with the Cepstral voice. Then we use that audio file for our purposes. The use is short term. That's pretty much it. Would like to move this functionality to Freeswitch. We don't need simultaneous streams as the volume of creating TS files is very low. We will worry about simultaneous streams in the future if it becomes necessary. Is mod_cepstral the way to go or is there a better updated way to get it done? Any suggestions will be greatly appreciated. Thanks. Mav

1 0

[Freeswitch-dev] sometimes some internal calls abandoned!
by S.Mohammad Emami Razavi 09 Oct '19

09 Oct '19

Hello everybody, Have you ever got `Abandoned` warning and your call hanged?! Sometimes casually our internal calls have this problem. Call is not bridged (like the device hangs!) and `Abandoned` warning logged. fs_cli console log tells: ``` 2019-10-09 09:01:47.502759 [WARNING] switch_core_state_machine.c:572 8a09055e-98b6-409a-9d01-8b1961860447 sofia/internal/138@***.***.***.***:5060 Abandoned ``` Best regards,

2 1

[Freeswitch-dev] Using different mrcp profiles in one SIP session
by Трусов Алексей 27 Sep '19

27 Sep '19

Hi everyone, The company I work for is developing a voice driven application that uses the FreeSwitch 1.6 and ESL to handle voice calls and voice recognition. During a single SIP session the app might use different MRCP profiles. The problem is that the MRCP profile name is appeared to be fixed for a single sip session. If you guys confirm that this issue exists I would like to merge the patch in the main trunk.

2 3

[Freeswitch-dev] Git repository moving to GitHub. IMPORTANT PLEASE READ
by Mike Jerris 24 Sep '19

24 Sep '19

By popular demand and after a bunch of hard work, we are thrilled to announce that the FreeSWITCH open source repository is now officially moved to GitHub! - All new issues, pull requests, and bug tracking will now happen at https://github.com/signalwire/freeswitch - All tooling for releases, package building, and testing has been switched to the new GitHub - Stash, Bitbucket, & Jira now in read-only status - We worked closely with GitHub to preserve the commit history - all commit SHAs have been retained to the beginning of time - If you have an open ticket in Jira that you'd like surfaced please re-open it via a GitHub Issue or PR We're very excited to get FreeSWITCH into a place where more developers feel comfortable contributing to the success of the project. Let us know if you have any questions!

1 0

[Freeswitch-dev] mod_com_g729 Codec license Transcoding not working
by Anand Sivaraj S 20 Sep '19

20 Sep '19

Hi, I've purchased 6 g729 codec license and activated it successfully. When I establish a call from my Service Provider ( using G729 ), call is received through the SIP Trunk. In FreeSwitch I have routed the call to an extension. When a sip client ( MircoSIP) logged into that extension ( configured for G.711 u-law and G.711 A-law) Call is not getting transferred and it returns currently not available. But When the sip client ( MircoSIP) is configured for G.711 u-law and G.711 A-law and G729 8kHz, then the call is going through. My concern is even after purchasing G729 license, Transcoding is not happening. If anyone has come across this problem and found a solution, your inputs are appreciated. Thank you. My Configuration values I the respective files are /usr/local/freeswitch/ conf/sip_profiles/external/EXTSIPGW.xml : <include> <gateway name=" EXTSIPGW "> <param name="proxy" value="192.168.168.34:5060"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="username" value="not-used"/> <param name="password" value="not-used"/> <param name="contact-params" value="tport=tcp"/> <param name="register-transport" value="tcp"/> </gateway> </include> /usr/local/freeswitch/ conf/vars.xml : <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,G729,H264,VP8"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,G722,PCMU,PCMA,G729,H264,VP8"/> /usr/local/freeswitch/ conf/autoload_configs/modules.conf.xml :  <load module="mod_spandsp"/> <load module="mod_g723_1"/>  <load module="mod_com_g729"/> <load module="mod_amr"/>   <load module="mod_b64"/>   <load module="mod_opus"/> fs_cli: Log freeswitch@debian> g729_info G.729A license available => 6 G.729A license allocated => 0 Regards, Anand Sivaraj S

2 1

[Freeswitch-dev] Support for getDisplayMedia API in Verto JS
by Erwin Eggenberger 27 Aug '19

27 Aug '19

Hi, I'd like to ask if support for the new getDisplayMedia API is already planned for the Verto JavaScript libraries, or if you're accepting code contributions adding support for it. We've integrated Verto and Freeswitch in our webinar solution, and as many of our customers use the screensharing feature, we'd like to switch to the getDisplayMedia API since Chrome and Firefox now both support it. Regards, Erwin -- Erwin Eggenberger /Senior Web Developer and Architect/ *Techcast GmbH* p: +49 (0) 89 356 477 304 a: Heßstraße 48b, 80798 München w: www.techcast.com <http://www.techcast.com> e: eeggenberger(a)techcast.com <mailto:eeggenberger@techcast.com> Digitalisierung vorantreiben: Techcast Newsletter abonnieren > <https://www.techcast.com/de/perspektiven-fuer-ihre-digitale-strategie> Amtsgericht München, HRB 178080, GF: Rosemarie Schuster

1 0

[Freeswitch-dev] mod_janus Pull Request merge request
by Richard Screene 16 Aug '19

16 Aug '19

Hello, Can I ask someone to merge this PR (https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/169… <https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/169…>)? The new endpoint module allows for FS to interface with the Janus audio bridge module. Mike Jerris did review it a couple of months ago and I made the suggested changes. But, it has not been merged. FYI, I have written some documentation at https://freeswitch.org/confluence/display/FREESWITCH/mod_janus <https://freeswitch.org/confluence/display/FREESWITCH/mod_janus> Many thanks, Richard

1 0

[Freeswitch-dev] Question about Jitter Buffer
by Dean Russ 12 Aug '19

12 Aug '19

Hello, I am trying to find out if there is a difference of using the jitter buffer setting 1P:5P instead of 20:100. They both appear to do the same thing. I'm also curious to know if anyone has ran into odd audio quality issues when running a large number of calls on a server with jitter buffer enabled on every call. freeswitch@deanfs uuid_jitterbuffer 0deee7e2-9e18-44a3-b353-198413409092 1p:5p +OK Success 0deee7e2 2019-08-11 19:56:01.729372 [DEBUG] switch_core_media.c:2549 Setting Jitterbuffer to 20ms (1 frames) (5 max frames) freeswitch@deanfs uuid_jitterbuffer 0deee7e2-9e18-44a3-b353-198413409092 20:100 +OK Success 0deee7e2 2019-08-11 19:59:27.869379 [DEBUG] switch_core_media.c:2549 Setting Jitterbuffer to 20ms (1 frames) (5 max frames) Thanks, Dean

1 0

[Freeswitch-dev] mod_janus Pull Request merge request
by Richard Screene 05 Aug '19

05 Aug '19

Hello, Can I ask someone to merge this PR (https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/169… <https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/169…>)? The new endpoint module allows for FS to interface with the Janus audio bridge module. Mike Jerris did review it a couple of months ago and I made the suggested changes. But, it has not been merged. FYI, I have written some documentation at https://freeswitch.org/confluence/display/FREESWITCH/mod_janus <https://freeswitch.org/confluence/display/FREESWITCH/mod_janus> Many thanks, Richard

1 0

[Freeswitch-dev] Regards to FreeSwitch STUNT and TURN Server setup
by HEMANTH REDDY 23 Jul '19

23 Jul '19

Hello Team, I'm facing the problem to integrate Freeswitch with STUNT and TURN server ,Could you please help to solve the problem. I'm sharing my error log in the below attachment. Thanks & Regards, Hemanth K

2 1