mirror of
https://github.com/RangeNetworks/openbts.git
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c0a5c1509e
git-svn-id: http://wush.net/svn/range/software/public/openbts/trunk@2242 19bc5d8c-e614-43d4-8b26-e1612bc8e597
92 lines
3.6 KiB
Plaintext
92 lines
3.6 KiB
Plaintext
[general]
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bindport=5060 ; asterisk 1.6
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; UDP Port to bind to (SIP standard port for unencrypted UDP
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; and TCP sessions is 5060)
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; bindport is the local UDP port that Asterisk will listen on
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bindaddr=0.0.0.0 ; asterisk 1.6
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; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
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; You can specify port here too, like 123.123.123.123:5080
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udpbindaddr=0.0.0.0 ; asterisk 1.8
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; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
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; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
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tos_sip=cs3 ; Sets TOS for SIP packets.
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tos_audio=ef ; Sets TOS for RTP audio packets.
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tos_video=af41 ; Sets TOS for RTP video packets.
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tos_text=af41 ; Sets TOS for RTP text packets.
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cos_sip=3 ; Sets 802.1p priority for SIP packets.
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cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
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cos_video=4 ; Sets 802.1p priority for RTP video packets.
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cos_text=3 ; Sets 802.1p priority for RTP text packets.
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maxexpiry=3600 ; Maximum allowed time of incoming registrations
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; and subscriptions (seconds)
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minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
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defaultexpiry=3600 ; Default length of incoming/outgoing registration
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dynamic_exclude_static=yes ; Disallow all dynamic hosts from registering
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; as any IP address used for staticly defined
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; hosts. This helps avoid the configuration
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; error of allowing your users to register at
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; the same address as a SIP provider.
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use_q850_reason=yes ; Set to yes add Reason header and use Reason header if it is available.
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;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
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; Defaults to 100 ms
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;timert1=500 ; Default T1 timer
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; Defaults to 500 ms or the measured round-trip
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; time to a peer (qualify=yes).
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;timerb=32000 ; Call setup timer. If a provisional response is not received
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; in this amount of time, the call will autocongest
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; Defaults to 64*timert1
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rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
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; on the audio channel
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; when we're not on hold. This is to be able to hangup
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; a call in the case of a phone disappearing from the net,
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; like a powerloss or grandma tripping over a cable.
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rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
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; on the audio channel
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; when we're on hold (must be > rtptimeout)
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;allowguest=no ; Allow or reject guest calls (default is yes)
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autocreatepeer=yes ; The Autocreatepeer option allows,
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; if set to Yes, any SIP ua to register with your Asterisk PBX as a peer.
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; This peer's settings will be based on global options.
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; The peer's name will be based on the user part of the Contact: header field's URL.
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context=phones ; Default context for incoming calls
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allowoverlap=no ; Disable overlap dialing support. (Default is yes)
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disallow=all ; need to disallow=all before we can use allow=
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allow=gsm ; GSM
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allow=ulaw ; ISDN US
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allow=alaw ; ISDN EU
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relaxdtmf=yes ; Relax dtmf handling
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dtmfmode=auto ; Set default dtmfmode for sending DTMF. Default: rfc2833
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; Other options:
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; info : SIP INFO messages (application/dtmf-relay)
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; shortinfo : SIP INFO messages (application/dtmf)
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; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
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; auto : Use rfc2833 if offered, inband otherwise
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; Zoiper is used as a fixture for factory testing.
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[zoiper]
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secret=3078923984
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callerid=2101
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canreinvite=no
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type=friend
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context=sip-local
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host=dynamic
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dtmfmode=auto
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; This is a test SIM provided with the BTS.
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[IMSI001010000000000]
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callerid=2100
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canreinvite=no
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type=friend
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context=sip-local
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host=dynamic
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