April 2020 Archives by author

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Starting: Wed Apr 1 12:13:33 UTC 2020
Ending: Thu Apr 30 14:41:46 UTC 2020
Messages: 134

• [Freeswitch-users] Device / extension state (status) functionality at Freeswitch or get User (Peer) / Gateway (Trunk) state   CpServiceSPb .
• [Freeswitch-users] Device / extension state (status) functionality at Freeswitch or get User (Peer) / Gateway (Trunk) state   CpServiceSPb .
• [Freeswitch-users] execute_on_answer equivalent for A-leg or bridged call?   Vallimamod Abdullah
• [Freeswitch-users] Need help with mod_v8 install   Aleksandar Bilanovic
• [Freeswitch-users] mod_conference and Hebrew-language conferencing (pro-bono production server for school system)   Yisroel Meir Blumstein
• [Freeswitch-users] How to connect SIP call to rtsp stream   Victor Chukalovskiy
• [Freeswitch-users] How to connect SIP call to rtsp stream   Victor Chukalovskiy
• [Freeswitch-users] Problems with TLS after upgrading to Buster   Victor Chukalovskiy
• [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?   John Covici
• [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?   John Covici
• [Freeswitch-users] send_dtmf only sends last 2 digits   John Covici
• [Freeswitch-users] Announcing SaraPhone, SIP WebRTC Open Source business phone   John Covici
• [Freeswitch-users] Announcing SaraPhone, SIP WebRTC Open Source business phone   John Covici
• [Freeswitch-users] SIP 302 Redirect   Loi Dang
• [Freeswitch-users] SIP 302 Redirect   Loi Dang
• [Freeswitch-users] SIP 302 Redirect   Loi Dang
• [Freeswitch-users] sofia reject MESSAGE method   Loi Dang
• [Freeswitch-users] sofia reject MESSAGE method   Loi Dang
• [Freeswitch-users] Stream File on both leg simultaneously   Jay Desai
• [Freeswitch-users] MySQL setup   Sean Devoy
• [Freeswitch-users] How to connect SIP call to rtsp stream   Seven Du
• [Freeswitch-users] Mod_conference with many participants. Question for experts   Seven Du
• [Freeswitch-users] curl zombie_exec   Seven Du
• [Freeswitch-users] Announcing SaraPhone, SIP WebRTC Open Source business phone   Seven Du
• [Freeswitch-users] freeswitch video conference recording black screen   Seven Du
• [Freeswitch-users] Answer ringing phone with uuid_phone_event behind load balancer   Jose Fco. Irles Durá
• [Freeswitch-users] send message via python script using ESL library   Youssef ELOUAM
• [Freeswitch-users] Stream File on both leg simultaneously   Carlos Eduardo
• [Freeswitch-users] Error on loading ESL.so on debian 10   Jose Figueroa
• [Freeswitch-users] execute_on_answer equivalent for A-leg or bridged call?   Thilo-Alexander Ginkel
• [Freeswitch-users] execute_on_answer equivalent for A-leg or bridged call?   Thilo-Alexander Ginkel
• [Freeswitch-users] SIP 302 Redirect   Giri
• [Freeswitch-users] SIP 302 Redirect   Giri
• [Freeswitch-users] SIP 302 Redirect   Giri
• [Freeswitch-users] SIP 302 Redirect   Giri
• [Freeswitch-users] Mod avmd   Piotr Gregor
• [Freeswitch-users] Mod avmd   Piotr Gregor
• [Freeswitch-users] mod_conference and Hebrew-language conferencing (pro-bono production server for school system)   Moishe Grunstein
• [Freeswitch-users] curl zombie_exec   Marcel Haldemann
• [Freeswitch-users] How to connect SIP call to rtsp stream   Faisal Hanif
• [Freeswitch-users] Calls per gateway   Faisal Hanif
• [Freeswitch-users] sofia reject MESSAGE method   Sean Hansen
• [Freeswitch-users] sofia reject MESSAGE method   Sean Hansen
• [Freeswitch-users] Rejecting calls (for a missed call service)   Sam van Herwaarden
• [Freeswitch-users] Freeswitch drops calls after 32 seconds   David Horton
• [Freeswitch-users] MySQL setup   Jurijs Ivolga
• [Freeswitch-users] freeswitch video conference recording black screen   Aidar Kamalov
• [Freeswitch-users] freeswitch video conference recording black screen   Aidar Kamalov
• [Freeswitch-users] send_dtmf only sends last 2 digits   Pete Kay
• [Freeswitch-users] Need help with mod_v8 install   Pete Kay
• [Freeswitch-users] Freeswitch drops calls after 32 seconds   Sonny Lafuente
• Auto Changing RTP concern   Sonny Lafuente
• [Freeswitch-users] Calls per gateway   Francesco Facco de Lagarda
• [Freeswitch-users] Calls per gateway   Francesco Facco de Lagarda
• [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?   Mitul Limbani
• [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?   Mitul Limbani
• [Freeswitch-users] Announcing SaraPhone, SIP WebRTC Open Source business phone   Mitul Limbani
• [Freeswitch-users] FYI - new Amazon Web Services TTS option   Avi Marcus
• [Freeswitch-users] How to play a file to conference moderator?   Avi Marcus
• [Freeswitch-users] Calls per gateway   Avi Marcus
• [Freeswitch-users] Calls per gateway   Avi Marcus
• [Freeswitch-users] Calls per gateway   Avi Marcus
• [Freeswitch-users] ADD 2 ICE CANDIDATES mod_verto   Jazmin Marina Florez Marino
• [Freeswitch-users] ADD 2 ICE CANDIDATES mod_verto   Jazmin Marina Florez Marino
• [Freeswitch-users] ADD 2 ICE CANDIDATES mod_verto   Jazmin Marina Florez Marino
• [Freeswitch-users] Changing RTP ports mid call   Joli Martinez
• [Freeswitch-users] unable to post   Joli Martinez
• [Freeswitch-users] Voice Conference Voice Choppy   Giovanni Maruzzelli
• [Freeswitch-users] iOS push notifications   Giovanni Maruzzelli
• [Freeswitch-users] iOS push notifications   Giovanni Maruzzelli
• [Freeswitch-users] Voice Conference Voice   Giovanni Maruzzelli
• [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?   Giovanni Maruzzelli
• [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?   Giovanni Maruzzelli
• [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?   Giovanni Maruzzelli
• [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?   Giovanni Maruzzelli
• [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?   Giovanni Maruzzelli
• [Freeswitch-users] Announcing SaraPhone, SIP WebRTC Open Source business phone   Giovanni Maruzzelli
• [Freeswitch-users] verto and conference for webrtc audio mcu   Giovanni Maruzzelli
• [Freeswitch-users] Announcing SaraPhone, SIP WebRTC Open Source business phone   Giovanni Maruzzelli
• [Freeswitch-users] Announcing SaraPhone, SIP WebRTC Open Source business phone   Giovanni Maruzzelli
• [Freeswitch-users] Freeswitch drops calls after 32 seconds   MikeKulls
• [Freeswitch-users] unable to post   Per Modin
• [Freeswitch-users] mod_verto: subscribe events for specific accounts   Gregor Nanger
• [Freeswitch-users] Auth rejected after upgrade to 1.10   David P
• [Freeswitch-users] TURN TCP to reduce TURN cost   David P
• [Freeswitch-users] ICE TCP to reduce TURN cost   David P
• [Freeswitch-users] Announcing SaraPhone, SIP WebRTC Open Source business phone   Vishal Pai
• [Freeswitch-users] Mod avmd   Vishal Pai
• [Freeswitch-users] verto and conference for webrtc audio mcu   Roberto Della Pasqua
• [Freeswitch-users] Using channel variables as values to time of day attributes in conditions   Chad Phillips
• [Freeswitch-users] curl zombie_exec   Mai Duy Quang
• [Freeswitch-users] Freeswitch drops calls after 32 seconds   Ken Rice
• [Freeswitch-users] Announcing SaraPhone, SIP WebRTC Open Source business phone   Moshe Rosenberg
• [Freeswitch-users] Mod_conference with many participants. Question for experts   Gilles SAUVAIRE
• [Freeswitch-users] Device / extension state (status) functionality at Freeswitch or get User (Peer) / Gateway (Trunk) state   Gilles SAUVAIRE
• [Freeswitch-users] mod_conference and Hebrew-language conferencing (pro-bono production server for school system)   Sergey Safarov
• mod_verto: subscribe events for specific accounts   António Silva
• [Freeswitch-users] mod_verto: subscribe events for specific accounts   António Silva
• [Freeswitch-users] Join us for SIP Chronicles Live #1, featuring Giovanni Maruzzelli   Maxim Sobolev
• [Freeswitch-users] iOS push notifications   Nathan Stratton
• [Freeswitch-users] MySQL setup   Vlad Toma
• [Freeswitch-users] Accessing the Identity header in the dialplan   Alexandru Tripon
• [Freeswitch-users] Changing RTP ports mid call   Valli A. Vallimamod
• [Freeswitch-users] Wrong number of tries with originate_retries?   Valli A. Vallimamod
• [Freeswitch-users] SIP help   David Villasmil
• [Freeswitch-users] Voice Conference Voice   David Villasmil
• [Freeswitch-users] MySQL setup   David Villasmil
• [Freeswitch-users] MySQL setup   David Villasmil
• [Freeswitch-users] Freeswitch drops calls after 32 seconds   David Villasmil
• [Freeswitch-users] Call not connecting   David Villasmil
• [Freeswitch-users] execute_on_answer equivalent for A-leg or bridged call?   David Villasmil
• [Freeswitch-users] Mod_conference with many participants. Question for experts   David Villasmil
• [Freeswitch-users] hangup_hook on an originated leg   David Villasmil
• [Freeswitch-users] hangup_hook on an originated leg   David Villasmil
• [Freeswitch-users] unable to post   David Villasmil
• [Freeswitch-users] Announcing SaraPhone, SIP WebRTC Open Source business phone   David Villasmil
• [Freeswitch-users] Add body to INFO reply   David Villasmil
• [Freeswitch-users] Add body to INFO reply   David Villasmil
• [Freeswitch-users] Wrong number of tries with originate_retries?   René Weiss
• [Freeswitch-users] SIP help   Brian West
• [Freeswitch-users] execute_on_answer equivalent for A-leg or bridged call?   Brian West
• [Freeswitch-users] Using channel variables as values to time of day attributes in conditions   Brian West
• [Freeswitch-users] sofia reject MESSAGE method   Brian West
• [Freeswitch-users] ADD 2 ICE CANDIDATES mod_verto   Brian West
• [Freeswitch-users] TURN TCP to reduce TURN cost   Brian West
• [Freeswitch-users] bridge two outbound calls but got different sdp   Leo Wolf
• [Freeswitch-users] Acl to ignore mdns ice candidates generated by browsers   Babak Yakhchali
• [Freeswitch-users] Voice Conference Voice Choppy   Ahmed habiba
• [Freeswitch-users] Voice Conference Voice Choppy(Ahabiba)   Ahmed habiba
• [Freeswitch-users] Voice Conference Voice   Ahmed habiba
• [Freeswitch-users] FreeSwitch for WebRTC to SIP/RTP gateway?   mickael.bride at orange.com
• [Freeswitch-users] SIP help   stephen at picardogroup.com
• [Freeswitch-users] PoP Flowroute   stephen at picardogroup.com
• [Freeswitch-users] Call not connecting   kumar uppu

Last message date: Thu Apr 30 14:41:46 UTC 2020
Archived on: Fri May 1 20:42:22 UTC 2020

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