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authorkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2005-10-04 22:51:59 +0000
committerkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2005-10-04 22:51:59 +0000
commit024f2617d8262e60fa1ee1a6496b079557fe72be (patch)
tree857ef7f7e70edb6af3ea2ed39635465b5625521a /configs/iax.conf.sample
parent28ee0af707a994129ce8cb8571f0c1349c616741 (diff)
make sample config files easier to ready (issue #5371)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6720 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs/iax.conf.sample')
-rwxr-xr-xconfigs/iax.conf.sample93
1 files changed, 47 insertions, 46 deletions
diff --git a/configs/iax.conf.sample b/configs/iax.conf.sample
index 0ebcdc83a..3fb5d338a 100755
--- a/configs/iax.conf.sample
+++ b/configs/iax.conf.sample
@@ -74,11 +74,11 @@ disallow=lpc10 ; Icky sound quality... Mr. Roboto.
; The jitter buffer's function is to compensate for varying
; network delay.
;
-; There are presently two jitterbuffer implementations available for * and chan_iax2;
-; the classic and the new, channel/application independent implementation. These
-; are controlled at compile-time. The new jitterbuffer additionally has support for PLC
-; which greatly improves quality as the jitterbuffer adapts size, and in compensating for lost
-; packets.
+; There are presently two jitterbuffer implementations available for Asterisk
+; and chan_iax2; the classic and the new, channel/application independent
+; implementation. These are controlled at compile-time. The new jitterbuffer
+; additionally has support for PLC which greatly improves quality as the
+; jitterbuffer adapts size, and in compensating for lost packets.
;
; All the jitter buffer settings except dropcount are in milliseconds.
; The jitter buffer works for INCOMING audio - the outbound audio
@@ -90,7 +90,8 @@ disallow=lpc10 ; Icky sound quality... Mr. Roboto.
; forcejitterbuffer=yes|no: in the ideal world, when we bridge VoIP channels
; we don't want to do jitterbuffering on the switch, since the endpoints
; can each handle this. However, some endpoints may have poor jitterbuffers
-; themselves, so this option will force * to always jitterbuffer, even in this case.
+; themselves, so this option will force * to always jitterbuffer, even in this
+; case.
; [This option presently applies only to the new jitterbuffer implementation]
;
; dropcount: the jitter buffer is sized such that no more than "dropcount"
@@ -105,15 +106,17 @@ disallow=lpc10 ; Icky sound quality... Mr. Roboto.
;
; resyncthreshold: when the jitterbuffer notices a significant change in delay
; that continues over a few frames, it will resync, assuming that the change in
-; delay was caused by a timestamping mix-up. The threshold for noticing a change
-; in delay is measured as twice the measured jitter plus this resync threshold.
-; Resycning can be disabled by setting this parameter to -1.
+; delay was caused by a timestamping mix-up. The threshold for noticing a
+; change in delay is measured as twice the measured jitter plus this resync
+; threshold.
+; Resyncing can be disabled by setting this parameter to -1.
; [This option presently applies only to the new jitterbuffer implementation]
;
-; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer should
-; return in a row. Since some clients do not send CNG/DTX frames to indicate
-; silence, the jitterbuffer will assume silence has begun after returning this
-; many interpolations. This prevents interpolating throughout a long silence.
+; maxjitterinterps: the maximum number of interpolation frames the jitterbuffer
+; should return in a row. Since some clients do not send CNG/DTX frames to
+; indicate silence, the jitterbuffer will assume silence has begun after
+; returning this many interpolations. This prevents interpolating throughout
+; a long silence.
; [This option presently applies only to the new jitterbuffer implementation]
;
; maxexcessbuffer: If conditions improve after a period of high jitter,
@@ -147,11 +150,11 @@ forcejitterbuffer=no
;trunkfreq=20 ; How frequently to send trunk msgs (in ms)
; Should we send timestamps for the individual sub-frames within trunk frames?
-; There is a small bandwidth use for these (less than 1kbps/call), but they ensure
-; that frame timestamps get sent end-to-end properly. If both ends of all your trunks
-; go directly to TDM, _and_ your trunkfreq equals the frame length for your codecs, you
-; can probably suppress these. The receiver must also support this feature, although
-; they do not also need to have it enabled.
+; There is a small bandwidth use for these (less than 1kbps/call), but they
+; ensure that frame timestamps get sent end-to-end properly. If both ends of
+; all your trunks go directly to TDM, _and_ your trunkfreq equals the frame
+; length for your codecs, you can probably suppress these. The receiver must
+; also support this feature, although they do not also need to have it enabled.
;
; trunktimestamps=yes
;
@@ -217,22 +220,21 @@ tos=lowdelay
;
;mailboxdetail=yes
;
-; If regcontext is specified, Asterisk will dynamically
-; create and destroy a NoOp priority 1 extension for a given
-; peer who registers or unregisters with us. The actual extension
-; is the 'regexten' parameter of the registering peer or its
-; name if 'regexten' is not provided. More than one regexten may be supplied
-; if they are separated by '&'. Patterns may be used in regexten.
+; If regcontext is specified, Asterisk will dynamically create and destroy
+; a NoOp priority 1 extension for a given peer who registers or unregisters
+; with us. The actual extension is the 'regexten' parameter of the registering
+; peer or its name if 'regexten' is not provided. More than one regexten
+; may be supplied if they are separated by '&'. Patterns may be used in
+; regexten.
;
;regcontext=iaxregistrations
;
-; If we don't get ACK to our NEW within 2000ms, and autokill is set
-; to yes, then we cancel the whole thing (that's enough time for one
-; retransmission only). This is used to keep things from stalling for a long
-; time for a host that is not available, but would be ill advised for bad
-; connections. In addition to 'yes' or 'no' you can also specify a number
-; of milliseconds. See 'qualify' for individual peers to turn on for just
-; a specific peer.
+; If we don't get ACK to our NEW within 2000ms, and autokill is set to yes,
+; then we cancel the whole thing (that's enough time for one retransmission
+; only). This is used to keep things from stalling for a long time for a host
+; that is not available, but would be ill advised for bad connections. In
+; addition to 'yes' or 'no' you can also specify a number of milliseconds.
+; See 'qualify' for individual peers to turn on for just a specific peer.
;
autokill=yes
;
@@ -274,8 +276,8 @@ autokill=yes
; has expired based on its registration interval, used the stored
; address information regardless. (yes|no)
-; Guest sections for unauthenticated connection attempts. Just
-; specify an empty secret, or provide no secret section.
+; Guest sections for unauthenticated connection attempts. Just specify an
+; empty secret, or provide no secret section.
;
[guest]
type=user
@@ -310,14 +312,13 @@ inkeys=freeworlddialup
;context=dundi-e164-local
;
-; Further user sections may be added, specifying a context and a
-; secret used for connections with that given authentication name.
-; Limited IP based access control is allowed by use of "allow" and
-; "deny" keywords. Multiple rules are permitted. Multiple permitted
-; contexts may be specified, in which case the first will be the default.
-; You can also override caller*ID so that when you receive a call you
-; set the Caller*ID to be what you want instead of trusting what
-; the remote user provides
+; Further user sections may be added, specifying a context and a secret used
+; for connections with that given authentication name. Limited IP based
+; access control is allowed by use of "allow" and "deny" keywords. Multiple
+; rules are permitted. Multiple permitted contexts may be specified, in
+; which case the first will be the default. You can also override caller*ID
+; so that when you receive a call you set the Caller*ID to be what you want
+; instead of trusting what the remote user provides
;
; There are three authentication methods that are supported: md5, plaintext,
; and rsa. The least secure is "plaintext", which sends passwords cleartext
@@ -372,11 +373,10 @@ host=216.207.245.47
;jitterbuffer=no ; Turn off jitter buffer for this peer
;
-; Peers can remotely register as well, so that they can be
-; mobile. Default IP's can also optionally be given but
-; are not required. Caller*ID can be suggested to the other
-; side as well if it is for example a phone instead of another
-; PBX.
+; Peers can remotely register as well, so that they can be mobile. Default
+; IP's can also optionally be given but are not required. Caller*ID can be
+; suggested to the other side as well if it is for example a phone instead of
+; another PBX.
;
;[dynamichost]
@@ -410,3 +410,4 @@ host=216.207.245.47
;secret=moofoo
;context=default
;permit=0.0.0.0/0.0.0.0
+