- 15 10月, 2010 15 次提交
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由 cypromis 提交于
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由 cypromis 提交于
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由 cypromis 提交于
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由 cypromis 提交于
PLEASE NEVER CHANGE ANYTHING IN THE DEBIAN DIRECTORY YOURSELF. ALWAYS CREATE A JIRA FIRST AND WAIT FOR THE CURRENT MAINTAINER TO DO THE CHANGES
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由 cypromis 提交于
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由 cypromis 提交于
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由 Jeff Lenk 提交于
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由 Marc Olivier Chouinard 提交于
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由 Michal Bielicki 提交于
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由 Michal Bielicki 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
controls are now bound to each member separately based on conference_controls channel var, then the caller-controls param in the profile or a default to "default" please test and report any issues in jira http://jira.freeswitch.org
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由 Anthony Minessale 提交于
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- 14 10月, 2010 16 次提交
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由 Michal Bielicki 提交于
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由 Jeff Lenk 提交于
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由 Michal Bielicki 提交于
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由 Michal Bielicki 提交于
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由 Michal Bielicki 提交于
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由 Michal Bielicki 提交于
updated modules in .debs to sync with .spec. left out flite since it requres more than 384mb to build debs with.
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由 Michal Bielicki 提交于
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由 Moises Silva 提交于
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由 Jeff Lenk 提交于
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由 David Yat Sin 提交于
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由 David Yat Sin 提交于
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由 David Yat Sin 提交于
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由 David Yat Sin 提交于
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由 Brian West 提交于
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由 Anthony Minessale 提交于
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由 Anthony Minessale 提交于
channel_variable: sdp_m_per_ptime Adds a new m= line for each distinct ptime in codec list. When this variable is not set: When mixing codecs with various ptime in a codec list, they will now be allowed to co-exist in the sdp but it will send no ptime attr. This means the ptime preferences on the offer will be ignored when mixing codecs with various ptimes. When receiving a codec list with no ptime attr, the ptime will be chosen from local preference instead of assuming 20ms This means if offer contains PCMU with not ptime and FS has PCMU@40i Dynamic payloads will now start at 98 and increment per additional dynamic codec per call. So now you can add CELT@32000h,CELT@48000h and each one will be auto-assigned a dynamic paylaod type.
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- 13 10月, 2010 9 次提交
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由 Michael S Collins 提交于
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由 Moises Silva 提交于
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由 Anthony Minessale 提交于
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由 Brian West 提交于
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由 Anthony Minessale 提交于
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由 David Yat Sin 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 Brian West 提交于
FS-2776: FS uses IPv6 under Proxy mode, and SIP Phone uses 6to4 tunneling IPv6 address, cause hearing nothing.
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