May 2019 Archives by date

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Starting: Tue May 7 22:07:30 CDT 2019
Ending: Fri May 31 13:25:24 CDT 2019
Messages: 54

• [asterisk-dev] WebRTC SFU: support add video track dynamically   Xiemin Chen
• [asterisk-dev] WebRTC SFU: support add video track dynamically   Joshua C. Colp
• [asterisk-dev] WebRTC SFU: support add video track dynamically   Xiemin Chen
• [asterisk-dev] WebRTC SFU: support add video track dynamically   Joshua C. Colp
• [asterisk-dev] WebRTC SFU: support add video track dynamically   Xiemin Chen
• [asterisk-dev] WebRTC SFU: support add video track dynamically   Joshua C. Colp
• [asterisk-dev] WebRTC SFU: support add video track dynamically   Xiemin Chen
• [asterisk-dev] WebRTC SFU: support add video track dynamically   Joshua C. Colp
• [asterisk-dev] WebRTC SFU: support add video track dynamically   Xiemin Chen
• [asterisk-dev] WebRTC SFU: support add video track dynamically   Joshua C. Colp
• [asterisk-dev] Asterisk 17 Feature Freeze Announcement   Matt Fredrickson
• [asterisk-dev] Audio to/from Asterisk   Sylvain Boily
• [asterisk-dev] Audio to/from Asterisk   Matt Fredrickson
• [asterisk-dev] Audio to/from Asterisk   Sylvain Boily
• [asterisk-dev] Manipulating Connected Line with ARI   Jean Aunis
• [asterisk-dev] Asterisk 13.27.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 16.4.0-rc1 Now Available   Asterisk Development Team
• [asterisk-dev] Manipulating Connected Line with ARI   Joshua C. Colp
• [asterisk-dev] pjsip asterisk 13.24: sips / srtp and Deutsche Telekom doesn't work because of missing mediasec parameters   Michael Maier
• [asterisk-dev] pjsip asterisk 13.24: sips / srtp and Deutsche Telekom doesn't work because of missing mediasec parameters   Joshua C. Colp
• [asterisk-dev] pjsip / asterisk 14+: Sometimes outbound calls are forbidden by provider   Michael Maier
• [asterisk-dev] pjsip / asterisk 14+: Sometimes outbound calls are forbidden by provider   Joshua C. Colp
• [asterisk-dev] pjsip / asterisk 14+: Sometimes outbound calls are forbidden by provider   Michael Maier
• [asterisk-dev] pjsip / asterisk 14+: Sometimes outbound calls are forbidden by provider   Joshua C. Colp
• [asterisk-dev] pjsip / asterisk 14+: Sometimes outbound calls are forbidden by provider   Michael Maier
• [asterisk-dev] pjsip / asterisk 14+: Sometimes outbound calls are forbidden by provider   Joshua C. Colp
• [asterisk-dev] pjsip / asterisk 14+: Sometimes outbound calls are forbidden by provider   Michael Maier
• [asterisk-dev] pjsip / asterisk 14+: Sometimes outbound calls are forbidden by provider   Joshua C. Colp
• [asterisk-dev] pjsip / asterisk 14+: Sometimes outbound calls are forbidden by provider   Michael Maier
• [asterisk-dev] Unable to compile asterisk12 with MALLOC_DEBUG   bala murugan
• [asterisk-dev] Unable to compile asterisk12 with MALLOC_DEBUG   Jared Smith
• [asterisk-dev] Unable to compile asterisk12 with MALLOC_DEBUG   bala murugan
• [asterisk-dev] WebRTC SFU: support add video track dynamically   Xiemin Chen
• [asterisk-dev] pjsip asterisk 13.24: sips / srtp and Deutsche Telekom doesn't work because of missing mediasec parameters   Michael Maier
• [asterisk-dev] WebRTC SFU: support add video track dynamically   Joshua C. Colp
• [asterisk-dev] pjsip asterisk 13.24: sips / srtp and Deutsche Telekom doesn't work because of missing mediasec parameters   Joshua C. Colp
• [asterisk-dev] WebRTC SFU: support add video track dynamically   Xiemin Chen
• [asterisk-dev] WebRTC SFU: support add video track dynamically   Joshua C. Colp
• [asterisk-dev] Unable to compile asterisk with MALLOC_DEBUG   bala murugan
• [asterisk-dev] Unable to compile asterisk with MALLOC_DEBUG   Dennis Buteyn
• [asterisk-dev] Unable to compile asterisk with MALLOC_DEBUG   Joshua C. Colp
• [asterisk-dev] pjsip asterisk 13.24: sips / srtp and Deutsche Telekom doesn't work because of missing mediasec parameters   Joshua C. Colp
• [asterisk-dev] pjsip asterisk 13.24: sips / srtp and Deutsche Telekom doesn't work because of missing mediasec parameters   Michael Maier
• [asterisk-dev] pjsip asterisk 13.24: sips / srtp and Deutsche Telekom doesn't work because of missing mediasec parameters   Michael Maier
• [asterisk-dev] Asterisk 13.27.0 Now Available   Asterisk Development Team
• [asterisk-dev] Asterisk 16.4.0 Now Available   Asterisk Development Team
• [asterisk-dev] Remove sdp attribute   Learn&Use
• [asterisk-dev] pjsip asterisk 13.24: sips / srtp and Deutsche Telekom doesn't work because of missing mediasec parameters   Michael Maier
• [asterisk-dev] Remove sdp attribute   Michael Maier
• [asterisk-dev] pjsip asterisk 13.24: sips / srtp and Deutsche Telekom doesn't work because of missing mediasec parameters   Joshua C. Colp
• [asterisk-dev] Remove sdp attribute   Joshua C. Colp
• [asterisk-dev] pjsip asterisk 13.24: sips / srtp and Deutsche Telekom doesn't work because of missing mediasec parameters   Michael Maier
• [asterisk-dev] pjsip asterisk 13.24: sips / srtp and Deutsche Telekom doesn't work because of missing mediasec parameters   Joshua C. Colp
• [asterisk-dev] pjsip asterisk 13.24: sips / srtp and Deutsche Telekom doesn't work because of missing mediasec parameters   Michael Maier

Last message date: Fri May 31 13:25:24 CDT 2019
Archived on: Fri May 31 13:22:49 CDT 2019

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