- 07 10月, 2014 6 次提交
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由 Mike Jerris 提交于
* commit 'eaaf9468': FS-6897: uuid_send_info enhancement that allows setting the Content-Type of the SIP INFO message
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由 Mike Jerris 提交于
* commit '747322dc': Remove Contact header from BYE and CANCEL requests.
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由 Mike Jerris 提交于
Merge pull request #65 in FS/freeswitch from ~MBRANCA/freeswitch:bugfix/OPENZAP-220-ftmod_libpri-don-t-close-channel to master * commit '7ec7c920': OPENZAP-220 fix blocked into read and add cause for a correct hangup
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由 Chris Rienzo 提交于
switch_pgsql.c switch_pgsql_next_result_timed() was using switch_time_now() for start time and switch_micro_time_now() for current time. These are different time sources that may not be in sync and could cause the query to timeout prematurely.
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由 Matteo Brancaleoni 提交于
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由 Markus von Arx 提交于
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- 06 10月, 2014 2 次提交
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由 Anthony Minessale 提交于
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由 Anthony Minessale 提交于
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- 03 10月, 2014 10 次提交
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由 Anthony Minessale 提交于
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由 Michael Jerris 提交于
FS-6781: #resolve #comment lets change this to always do confirm to match the other place where we set this
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由 Anthony Minessale 提交于
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由 Michael Jerris 提交于
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由 Anthony Minessale 提交于
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由 Anthony Minessale 提交于
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由 Michael Jerris 提交于
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由 Anthony Minessale 提交于
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由 Jeff Lenk 提交于
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由 Jeff Lenk 提交于
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- 02 10月, 2014 8 次提交
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由 Anthony Minessale 提交于
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由 Michael Jerris 提交于
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由 Anthony Minessale 提交于
%FEATURE add bypass_media_resume_on_hold and bypass_media_after_hold variables to be set to true to enable these functions on a per channel basis
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由 Anthony Minessale 提交于
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由 Spencer Thomason 提交于
Per rfc3261 the Contact header is not applicable and MUST not appear in the request. FS-5868 #resolve
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由 Anthony Minessale 提交于
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由 Anthony Minessale 提交于
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由 Anthony Minessale 提交于
the other way works better revert 91ffe171 to use high quality on stereo calls
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- 01 10月, 2014 11 次提交
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由 Anthony Minessale 提交于
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由 Anthony Minessale 提交于
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由 Anthony Minessale 提交于
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由 Anthony Minessale 提交于
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由 Anthony Minessale 提交于
FS-6822 #comment The code in question appears to have been added by me (18f20e24). I think this patch is the correct solution.
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由 Jeff Lenk 提交于
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由 Michael Jerris 提交于
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由 Anthony Minessale 提交于
FS-6880 #resolve #comment I would think that in real life once the call agreed on a codec it would only offer the negotiated codecs but we can fix this to always filter for good measure. I am not sure what the ramifications are of filtering responses but I think this patch will do so as well.
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由 Brian West 提交于
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由 Jeff Lenk 提交于
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- 30 9月, 2014 3 次提交
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由 Brian West 提交于
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由 Anthony Minessale 提交于
VARIABLE: bypass_media_sdp_filter Can be set globally or per leg on the inbound side of a bypass_media bridge. VALID FILTERS: remove(): Removes the specified codec if it exists in the SDP. only(): Removes all codecs besides the one specified (providing that it exists in the sdp) (will not remove telephone-event)) EXAMPLE 1 (remove everything leaving only g729): <action application="set" data="bypass_media_sdp_filter=only(g729)"/> <action application="set" data="bypass_media=true"/> <action application="bridge" data="sofia/internal/1238@conference.freeswitch.org"/> EXAMPLE 2 (remove everything leaving only g729 and also remove dtmf): <action application="set" data="bypass_media_sdp_filter=only(g729)|remove(telephone-event)"/> <action application="set" data="bypass_media=true"/> <action application="bridge" data="sofia/internal/1238@conference.freeswitch.org"/> EXAMPLE 3 (remove alaw and speex): <action application="set" data="bypass_media_sdp_filter=remove(pcma)|remove(speex)"/> <action application="set" data="bypass_media=true"/> <action application="bridge" data="sofia/internal/1238@conference.freeswitch.org"/>
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由 Anthony Minessale 提交于
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