[asterisk-users] Asterisk 16.4.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk
16.4.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.4.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
New Features made in this release:
---
* ASTERISK-28375 - res_pjsip: New configuration setting to
allow disabling norefersub
(Reported by Dan Cropp)
* ASTERISK-28320 - Added ARI resource
/ari/channels/{channelid}/rtp_statistics
(Reported by
sungtae kim)
Bugs fixed in this release:
---
* ASTERISK-28427 - new mwi.h include missing from some dahdi
source files, causes build failure
(Reported by Guido
Falsi)
* ASTERISK-28412 - GCC 9 catches more string formatting issues
(Reported by George Joseph)
* ASTERISK-28379 - pjsip: show channelstats incorrect
information output
(Reported by Vyrva Igor)
* ASTERISK-28399 - channel.c: Exceptionally long queue length
queuing
(Reported by Abhay Gupta)
* ASTERISK-28392 - The no-partial-inlining flag isn't passed to
the bundled pjproject or jansson builds
(Reported by
George Joseph)
* ASTERISK-28402 - res_pjsip_registrar: SEGV in
registrar_find_contact
(Reported by Ross Beer)
* ASTERISK-27756 - bridge: Failure to impart a channel results
in bad data causing crash
(Reported by Abhay Gupta)
* ASTERISK-26718 - ARI: Bridge destroying doesn't work as
expected
(Reported by Marin Odrljin)
* ASTERISK-28143 - app_amd: Infinite loop on silent calls
(Reported by Abhay Gupta)
* ASTERISK-28353 - stasis: Crash at shutdown when statistics
enabled
(Reported by Joshua C. Colp)
* ASTERISK-28374 - latest asterisk unconditionally launch gcc
--version, even if the compiler is different
(Reported by
Guido Falsi)
* ASTERISK-28391 - res_indications: Crash requesting
autocomplete on indications cli command
(Reported by Lucas
Mendes)
* ASTERISK-27935 - app_voicemail: emailbody per user can't
contain commas
(Reported by Sébastien Duthil)
* ASTERISK-17695 - 1.8.3.2 extenpatternmatchnew=yes cannot find
extensions with '-' in them
(Reported by test011)
* ASTERISK-17799 - AEL reload causes loss of control in a
macro
(Reported by Kirill Katsnelson)
* ASTERISK-18593 - AEL for loops use Macro app and pipe
delimiter
(Reported by Luke-Jr)
* ASTERISK-14939 - AEL parsers does not find existing label
(Reported by klaus3000)
* ASTERISK-20182 - Parsing a label beginning with a numeric
character in all Goto/GotoIf/GotoIfTime application causes
unexpected behavior
(Reported by Janu)
* ASTERISK-28348 - Failed to initialize OOH323 endpoint-OOH323
Disabled
(Reported by Dmitry Shubin)
* ASTERISK-28371 - chan_pjsip: DTMF Mode auto_info fallback
lead to both inband and info
(Reported by Salah Ahmed)
* ASTERISK-28319 - musl: Crash on startup when loading modules
(Reported by Sebastian Kemper)
* ASTERISK-28362 - strtok_r() makes gcc compile warning
(Reported by sungtae kim)
* ASTERISK-28255 - res_rtp_asterisk: REMB RTCP packet sending
may be incorrect
(Reported by Joshua C. Colp)
Improvements made in this release:
---
* ASTERISK-28401 - app_confbridge: Add *_all remb behavior
variants
(Reported by Joshua C. Colp)
* ASTERISK-28400 - res_rtp_asterisk / res_pjsip_sdp_rtp: Add
support for transport-cc
(Reported by Joshua C. Colp)
* ASTERISK-28363 - Millisecond-resolution call stats including
PDD in channel variables
(Reported by Antoni Goldstein)
* ASTERISK-20207 - Asterisk should clear out any .lock files in
the voice mail directory on startup.
(Reported by Steven
Wheeler)
* ASTERISK-28111 - build: CHANGES/UPGRADE are irritating to
work with.
(Reported by Corey Farrell)
* ASTERISK-28343 - Added app_name, app_data to channel type
(Reported by sungtae kim)
* ASTERISK-28264 - Added topic_all container
(Reported by
sungtae kim)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
Thank you for your continued support of Asterisk!
--
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
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http://
[asterisk-users] Asterisk 13.27.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk
13.27.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.27.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
New Features made in this release:
---
* ASTERISK-28375 - res_pjsip: New configuration setting to
allow disabling norefersub
(Reported by Dan Cropp)
* ASTERISK-28320 - Added ARI resource
/ari/channels/{channelid}/rtp_statistics
(Reported by
sungtae kim)
Bugs fixed in this release:
---
* ASTERISK-28427 - new mwi.h include missing from some dahdi
source files, causes build failure
(Reported by Guido
Falsi)
* ASTERISK-28412 - GCC 9 catches more string formatting issues
(Reported by George Joseph)
* ASTERISK-28392 - The no-partial-inlining flag isn't passed to
the bundled pjproject or jansson builds
(Reported by
George Joseph)
* ASTERISK-28402 - res_pjsip_registrar: SEGV in
registrar_find_contact
(Reported by Ross Beer)
* ASTERISK-28143 - app_amd: Infinite loop on silent calls
(Reported by Abhay Gupta)
* ASTERISK-28353 - stasis: Crash at shutdown when statistics
enabled
(Reported by Joshua C. Colp)
* ASTERISK-28374 - latest asterisk unconditionally launch gcc
--version, even if the compiler is different
(Reported by
Guido Falsi)
* ASTERISK-28391 - res_indications: Crash requesting
autocomplete on indications cli command
(Reported by Lucas
Mendes)
* ASTERISK-27935 - app_voicemail: emailbody per user can't
contain commas
(Reported by Sébastien Duthil)
* ASTERISK-17695 - 1.8.3.2 extenpatternmatchnew=yes cannot find
extensions with '-' in them
(Reported by test011)
* ASTERISK-17799 - AEL reload causes loss of control in a
macro
(Reported by Kirill Katsnelson)
* ASTERISK-18593 - AEL for loops use Macro app and pipe
delimiter
(Reported by Luke-Jr)
* ASTERISK-14939 - AEL parsers does not find existing label
(Reported by klaus3000)
* ASTERISK-20182 - Parsing a label beginning with a numeric
character in all Goto/GotoIf/GotoIfTime application causes
unexpected behavior
(Reported by Janu)
* ASTERISK-28348 - Failed to initialize OOH323 endpoint-OOH323
Disabled
(Reported by Dmitry Shubin)
* ASTERISK-28371 - chan_pjsip: DTMF Mode auto_info fallback
lead to both inband and info
(Reported by Salah Ahmed)
* ASTERISK-28362 - strtok_r() makes gcc compile warning
(Reported by sungtae kim)
Improvements made in this release:
---
* ASTERISK-28363 - Millisecond-resolution call stats including
PDD in channel variables
(Reported by Antoni Goldstein)
* ASTERISK-20207 - Asterisk should clear out any .lock files in
the voice mail directory on startup.
(Reported by Steven
Wheeler)
* ASTERISK-28111 - build: CHANGES/UPGRADE are irritating to
work with.
(Reported by Corey Farrell)
* ASTERISK-28343 - Added app_name, app_data to channel type
(Reported by sungtae kim)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.27.0
Thank you for your continued support of Asterisk!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.26.0 webRTC: Asterisk not passing along video
On Thu, May 30, 2019, at 11:30 AM, Jonas Kellens wrote: > Hello > > is this mailing list still active ? Seems like it. :D I responded previously. Many people have moved to Discourse[1] though and it sees more activity. [1] https://community.asterisk.org/ -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.26.0 webRTC: Asterisk not passing along video
Hello is this mailing list still active ? Op 10-05-19 om 14:10 schreef Jonas Kellens: Hello I am trying to set up webRTC video calls from my Chrome webbrowser (Fedora) to my Chrome webbrowser (Windows 10). There is local video input (I can see myself), but never video on the receiving side. This is the case in both directions (so it makes no difference which peer is calling which peer). Both webRTC SIP peers have opus and H264 codec in their peer definition : Video Support: Yes Prim.Transp. : WS Allowed.Trsp : WSS SIP Options : (none) Codecs : (opus|h264) Status : OK (75 ms) Useragent : SIP.js/0.12.0 Reg. Contact : sip:llghjqha@192.0.2.239;transport=wss RTP Engine : asterisk Encryption : Yes RTCP Mux : Yes Video Support: Yes Prim.Transp. : WS Allowed.Trsp : WSS SIP Options : (none) Codecs : (opus|h264) Status : OK (47 ms) Useragent : SIP.js/0.12.0 Reg. Contact : sip:6ltm4mqe@192.0.2.7;transport=wss RTP Engine : asterisk Encryption : Yes RTCP Mux : Yes In general sip.conf I have : videosupport=yes disallow=all allow=alaw allow=opus allow=h264 When one peer makes a SIP INVITE for a video call, it is clear to me that the necessary codec information is present (this all looks fine to me) : (calling webRTC client) SIP Debugging Enabled for IP: 99.99.255.55 [May 10 10:45:24] [May 10 10:45:24] <--- SIP read from WS:99.99.255.55:47732 ---> [May 10 10:45:24] INVITE sip:1...@wss.mydomain.tld SIP/2.0 [May 10 10:45:24] Via: SIP/2.0/WSS 192.0.2.7;branch=z9hG4bK9220692 [May 10 10:45:24] Max-Forwards: 70 [May 10 10:45:24] To: [May 10 10:45:24] From: "WC User Chrome" ;tag=sdmbqkquhe [May 10 10:45:24] Call-ID: 3g51uvbnnioje6riokqu [May 10 10:45:24] CSeq: 4132 INVITE [May 10 10:45:24] Contact: [May 10 10:45:24] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER [May 10 10:45:24] Supported: outbound [May 10 10:45:24] User-Agent: SIP.js/0.12.0 [May 10 10:45:24] Content-Type: application/sdp [May 10 10:45:24] Content-Length: 5098 [May 10 10:45:24] [May 10 10:45:24] v=0 [May 10 10:45:24] o=- 6075323372920596423 2 IN IP4 127.0.0.1 [May 10 10:45:24] s=- [May 10 10:45:24] t=0 0 [May 10 10:45:24] a=group:BUNDLE audio video [May 10 10:45:24] a=msid-semantic: WMS I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E [May 10 10:45:24] m=audio 34197 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 [May 10 10:45:24] c=IN IP4 99.99.255.55 [May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0 [May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 192.168.1.110 34197 typ host generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 34197 typ srflx raddr 192.168.1.110 rport 34197 generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 192.168.1.110 9 typ host tcptype active generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=ice-ufrag:y8md [May 10 10:45:24] a=ice-pwd:nyjEuDKhDVeu8B+OyvuEp6le [May 10 10:45:24] a=ice-options:trickle [May 10 10:45:24] a=fingerprint:sha-256 C9:33:B0:E9:7C:F4:F2:39:98:A6:5C:AE:16:7F:5E:18:99:8F:9F:EB:DC:C6:E3:D5:EA:5B:AE:CD:DE:75:79:0B [May 10 10:45:24] a=setup:actpass [May 10 10:45:24] a=mid:audio [May 10 10:45:24] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level [May 10 10:45:24] a=sendrecv [May 10 10:45:24] a=rtcp-mux [May 10 10:45:24] a=rtpmap:111 opus/48000/2 [May 10 10:45:24] a=rtcp-fb:111 transport-cc [May 10 10:45:24] a=fmtp:111 minptime=10;useinbandfec=1 [May 10 10:45:24] a=rtpmap:103 ISAC/16000 [May 10 10:45:24] a=rtpmap:104 ISAC/32000 [May 10 10:45:24] a=rtpmap:9 G722/8000 [May 10 10:45:24] a=rtpmap:0 PCMU/8000 [May 10 10:45:24] a=rtpmap:8 PCMA/8000 [May 10 10:45:24] a=rtpmap:106 CN/32000 [May 10 10:45:24] a=rtpmap:105 CN/16000 [May 10 10:45:24] a=rtpmap:13 CN/8000 [May 10 10:45:24] a=rtpmap:110 telephone-event/48000 [May 10 10:45:24] a=rtpmap:112 telephone-event/32000 [May 10 10:45:24] a=rtpmap:113 telephone-event/16000 [May 10 10:45:24] a=rtpmap:126 telephone-event/8000 [May 10 10:45:24] a=ssrc:401971016 cname:cd1IocMPYzY4lNYJ [May 10 10:45:24] a=ssrc:401971016 msid:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E f8eee8bd-dd47-4c14-866d-07069cab255f [May 10 10:45:24] a=ssrc:401971016 mslabel:I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E [May 10 10:45:24] a=ssrc:401971016 label:f8eee8bd-dd47-4c14-866d-07069cab255f [May 10 10:45:24] m=video 48086 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 123 127 122 125 107 108 109 124 [May 10 10:45:24] c=IN IP4 99.99.255.55 [May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0 [May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 192.168.1.110 48086 typ host generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:260925276 1 udp 1686052607 99.99.255.55 48086 typ srflx raddr 192.168.1.110 rport 48086 generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:3225853208 1 tcp 1518280447 192.168.1.110 9 typ
