inbound calls accepted by sip
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17
sip.conf
17
sip.conf
@@ -290,7 +290,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
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;mwiexpiry=3600 ; Expiry time for outgoing MWI subscriptions
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;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
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;maxforwards=70 ; Setting for the SIP Max-Forwards: header (loop prevention)
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; Default value is 70
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; Default value is 70
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;qualifyfreq=60 ; Qualification: How often to check for the host to be up in seconds
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qualifyfreq=20 ; Qualification: How often to check for the host to be up in seconds
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; and reported in milliseconds with sip show settings.
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; and reported in milliseconds with sip show settings.
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; Set to low value if you use low timeout for NAT of UDP sessions
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; Set to low value if you use low timeout for NAT of UDP sessions
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; Default: 60
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; Default: 60
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@@ -817,6 +817,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; Examples:
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; Examples:
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;
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;
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;register => 1234:password@mysipprovider.com
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;register => 1234:password@mysipprovider.com
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register => 7467:12zX22bH02q2@hg.eventphone.de
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;
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;
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; This will pass incoming calls to the 's' extension
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; This will pass incoming calls to the 's' extension
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;
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;
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@@ -1617,3 +1618,17 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
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; if the nat option is enabled. If a single RTP packet is received Asterisk will know the
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; external IP address of the remote device. If port forwarding is done at the client side
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; external IP address of the remote device. If port forwarding is done at the client side
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; then UDPTL will flow to the remote device.
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; then UDPTL will flow to the remote device.
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[eventphone]
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type=friend
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host=hg.eventphone.de
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defaultuser=7467
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secret=12zX22bH02q2
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context=from-eventphone
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qualify=yes
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disallow=all
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allow=alaw
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direct-media=no
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canrewrite=no
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dtmfmode=rfc2833
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insecure=invite,port
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