64 lines
2.0 KiB
Plaintext
64 lines
2.0 KiB
Plaintext
[general]
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# valid values for log level are: ERROR, WARN, NOTICE, INFO, DEBUG
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logLevel = NOTICE
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[sip]
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# list of valid SIP URIs for incoming connections separated by space
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# supported wildcards: *
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# if you want to allow calls from any URI, write *@*
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validUriExpression = *@sip.example.com *@127.0.0.1
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host = sip.example.org
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port = 5060
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user = mumsi
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password = foobar
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# length of single voice frame in ms. Valid values are 10, 20, 40, 60 ms.
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# Adjust it if you need to meet the specific bandwidth requirements of Murmur server
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frameLength = 40
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# Set the maximum number of SIP calls to allow simultaneously. This should be <= 32.
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# If you need more, recompile PJSUA LIB and also modify the define in main.hpp.
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max_calls = 1
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[mumble]
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host = example.org
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port = 64738
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user = mumsi
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password = foobar
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channelNameExpression =
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# When here is no SIP connection, the mumble state is set to self_mute/self_deaf
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# so the other users can easily see whether the SIP is connected even when not
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# in the same group
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autodeaf = 1
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# Bitrate of Opus encoder in B/s
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# Adjust it if you need to meet the specific bandwidth requirements of Murmur server
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opusEncoderBitrate = 16000
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# Set to 1 to use client certificates. The certs must be named <user>-cert.pem and
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# the private keys <user>-key.pem.
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use_certs = 0
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[app]
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# Caller PIN needed to authenticate the phone call itself. The caller presses
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# the PIN, followed by the hash '#' key. On success, the caller is
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# unmuted/undeafened. On failure, the SIP call is hung up.
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pin = 4321
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[files]
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# These files are used for the caller and mumble channel audio clips.
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# The paths below assume that you are running ./mumsi in the build/ dir.
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welcome = ../media/welcome.wav
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prompt_pin = ../media/prompt-pin.wav
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entering_channel = ../media/entering-channel.wav
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announce_new_caller = ../media/announce-new-caller.wav
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invalid_pin = ../media/invalid-pin.wav
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goodbye = ../media/goodbye.wav
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mute_on = ../media/mute-on.wav
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mute_off = ../media/mute-off.wav
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menu = ../media/menu.wav
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