/
sip.conf
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/
sip.conf
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; WARNING do not change this file, but instead use sip-custom-register.conf and sip-custom-contexts.conf
; as this will limit the amount of conflicts when upgrading
[general]
bindport=5060 ; asterisk 1.6
; UDP Port to bind to (SIP standard port for unencrypted UDP
; and TCP sessions is 5060)
; bindport is the local UDP port that Asterisk will listen on
bindaddr=0.0.0.0 ; asterisk 1.6
; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; You can specify port here too, like 123.123.123.123:5080
udpbindaddr=0.0.0.0 ; asterisk 1.8
; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
tos_sip=cs3 ; Sets TOS for SIP packets.
tos_audio=ef ; Sets TOS for RTP audio packets.
tos_video=af41 ; Sets TOS for RTP video packets.
tos_text=af41 ; Sets TOS for RTP text packets.
cos_sip=3 ; Sets 802.1p priority for SIP packets.
cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
cos_video=4 ; Sets 802.1p priority for RTP video packets.
cos_text=3 ; Sets 802.1p priority for RTP text packets.
maxexpiry=3600 ; Maximum allowed time of incoming registrations
; and subscriptions (seconds)
minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
defaultexpiry=3600 ; Default length of incoming/outgoing registration
dynamic_exclude_static=yes ; Disallow all dynamic hosts from registering
; as any IP address used for staticly defined
; hosts. This helps avoid the configuration
; error of allowing your users to register at
; the same address as a SIP provider.
use_q850_reason=yes ; Set to yes add Reason header and use Reason header if it is available.
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
; Defaults to 100 ms
;timert1=500 ; Default T1 timer
; Defaults to 500 ms or the measured round-trip
; time to a peer (qualify=yes).
;timerb=32000 ; Call setup timer. If a provisional response is not received
; in this amount of time, the call will autocongest
; Defaults to 64*timert1
rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
; on the audio channel
; when we're not on hold. This is to be able to hangup
; a call in the case of a phone disappearing from the net,
; like a powerloss or grandma tripping over a cable.
rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
; on the audio channel
; when we're on hold (must be > rtptimeout)
allowguest=no ; Allow or reject guest calls (default is yes)
autocreatepeer=no ; The Autocreatepeer option allows,
; if set to Yes, any SIP ua to register with your Asterisk PBX as a peer.
; This peer's settings will be based on global options.
; The peer's name will be based on the user part of the Contact: header field's URL.
context=from-openBTS ; Default context for incoming calls
;context=phones ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
disallow=all ; need to disallow=all before we can use allow=
allow=gsm ; GSM
allow=ulaw ; ISDN US
allow=alaw ; ISDN EU
relaxdtmf=yes ; Relax dtmf handling (only to be used for connecting to aserisk 1.2 and INDBAND)
dtmfmode=rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
; Other options:
; info : SIP INFO messages (application/dtmf-relay)
; shortinfo : SIP INFO messages (application/dtmf)
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
; auto : Use rfc2833 if offered, inband otherwise
canreinvite=no ; no reinvites from Asterisk
directmedia=no ; Asterisk by default tries to redirect the
; RTP media stream to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is behind a NAT).
; The default setting is YES. If you have all clients
; behind a NAT, or for some other reason want Asterisk to
; stay in the audio path, you may want to turn this off.
; This setting also affect direct RTP
; at call setup (a new feature in 1.4 - setting up the
; call directly between the endpoints instead of sending
; a re-INVITE).
callcounter=yes ; Enable call counters on devices. This can be set per
; device too.
#include sip-custom-register.conf
[CodecBTS](!)
disallow=all ; need to disallow=all before we can use allow
allow=gsm ; GSM
allow=ulaw ; ISDN US
allow=alaw ; ISDN EU
[optionsBTS](!)
type=peer
context=from-openBTS
dtmfmode=rfc2833
canreinvite=no
qualify=no ; openbts do not support OPTION
insecure=port,invite
;If you need to make any changes please add them to sip-custom-contexts.conf
#include sip-custom-contexts.conf