forked from Mirrors/freeswitch
fe09ec44e8
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@9551 d0543943-73ff-0310-b7d9-9358b9ac24b2
101 lines
5.1 KiB
XML
101 lines
5.1 KiB
XML
<!-- Preprocessor Variables
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These are introduced when configuration strings must be consistent across modules.
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-->
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<X-PRE-PROCESS cmd="set" data="sound_prefix=$${base_dir}/sounds/en/us/callie"/>
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<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
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<X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
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<!-- xmpp_client_profile and xmpp_server_profile
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xmpp_client_profile can be any string.
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xmpp_server_profile is appended to "dingaling_" to form the database name
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containing the "subscriptions" table.
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used by: dingaling.conf.xml enum.conf.xml
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-->
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<!--
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Examples of codec options: (module must be compiled and loaded)
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codecname[@8000h|16000h|32000h[@XXi]]
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XX is the frame size must be multples allowed for the codec
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FreeSWITCH can support 10-120ms on some codecs.
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We do not support exceeding the MTU of the RTP packet.
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iLBC@30i - iLBC using mode=30 which will win in all cases.
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DVI4@8000h@20i - IMA ADPCM 8kHz using 20ms ptime. (multiples of 10)
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DVI4@16000h@40i - IMA ADPCM 16kHz using 40ms ptime. (multiples of 10)
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speex@8000h@20i - Speex 8kHz using 20ms ptime.
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speex@16000h@20i - Speex 16kHz using 20ms ptime.
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speex@32000h@20i - Speex 32kHz using 20ms ptime.
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GSM@40i - GSM 8kHz using 40ms ptime. (GSM is done in multiples of 20, Default is 20ms)
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G722 - G722 16kHz using default 20ms ptime. (multiples of 10)
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PCMU - G711 8kHz ulaw using default 20ms ptime. (multiples of 10)
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PCMA - G711 8kHz alaw using default 20ms ptime. (multiples of 10)
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G726-16 - G726 16kbit adpcm using default 20ms ptime. (multiples of 10)
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G726-24 - G726 24kbit adpcm using default 20ms ptime. (multiples of 10)
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G726-32 - G726 32kbit adpcm using default 20ms ptime. (multiples of 10)
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G726-40 - G726 40kbit adpcm using default 20ms ptime. (multiples of 10)
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AAL2-G726-16 - Same as G726-16 but using AAL2 packing. (multiples of 10)
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AAL2-G726-24 - Same as G726-24 but using AAL2 packing. (multiples of 10)
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AAL2-G726-32 - Same as G726-32 but using AAL2 packing. (multiples of 10)
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AAL2-G726-40 - Same as G726-40 but using AAL2 packing. (multiples of 10)
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LPC - LPC10 using 90ms ptime (only supports 90ms at this time in FreeSWITCH)
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L16 - L16 isn't recommended for VoIP but you can do it. L16 can exceed the MTU rather quickly.
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These are the passthru audio codecs:
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G729 - G729 in passthru mode. (mod_g729)
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G723 - G723.1 in passthru mode. (mod_g723_1)
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AMR - AMR in passthru mode. (mod_amr)
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These are the passthru video codecs: (mod_h26x)
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H261 - H.261 Video
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H263 - H.263 Video
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H263-1998 - H.263-1998 Video
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H263-2000 - H.263-2000 Video
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H264 - H.264 Video
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-->
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<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM"/>
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<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
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<X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/>
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<X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/>
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<!-- bind_server_ip
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Can be an ip address, a dns name, or "auto".
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This determines an ip address available on this host to bind.
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If you are separating RTP and SIP traffic, you will want to have
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use different addresses where this variable appears.
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Used by: dingaling.conf.xml
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-->
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<X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
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<!-- external_rtp_ip
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Used as the public IP address for SDP.
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Can be an ip address or a string like "stun:stun.server.com"
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If unspecified, the bind_server_ip value is used.
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Used by: sofia.conf.xml dingaling.conf.xml
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-->
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<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
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<!-- external_sip_ip
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Used as the public IP address for SDP.
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Can be an ip address or a string like "stun:stun.server.com"
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If unspecified, the bind_server_ip value is used.
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Used by: sofia.conf.xml dingaling.conf.xml
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-->
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<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
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<!-- unroll-loops
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Used to turn on sip loopback unrolling.
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-->
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<X-PRE-PROCESS cmd="set" data="unroll_loops=true"/>
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<!-- outbound_caller_id and outbound_caller_name
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The caller ID telephone number we should use when calling out.
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Used by: conference.conf.xml
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-->
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<X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
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<X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
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<X-PRE-PROCESS cmd="set" data="call_debug=false"/>
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<X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2200,400,450)"/>
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<X-PRE-PROCESS cmd="set" data="us-ring=%(2000, 4000, 440.0, 480.0)"/>
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<X-PRE-PROCESS cmd="set" data="fr-ring=%(1500, 3500, 440.0, 0.0)"/>
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<X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/>
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