The account code can be defined on a per-user basis. If defined, this account code will be assigned to a call record whenever no specific user account code is set. The accountcode name configured will be used as the filename.csv in the /var/log/asterisk/cdr-csv/ directory to store Call Detail Records (CDRs) for the user/peer/friend:

accountcode=iax-username

Support for ADSI (Analog Display Services Interface) can be enabled if you have ADSI-compatible CPE equipment:

adsi=yes|no

Specific codecs can be allowed or disallowed, limiting codec use to those preferred by the system designer. allow and disallow can also be defined on a per-channel basis. Keep in mind that allow statements in the [general] section will carry over to each of the channels, unless you reset with a disallow=all. Codec negotiation is attempted in the order in which the codecs are defined. Best practice suggests that you define disallow=all, followed by explicit allow statements for each codec you wish to use. If nothing is defined, allow=all is assumed:

disallow=all
allow=ulaw
allow=gsm
allow=ilbc

Automatic Message Accounting (AMA) is defined in the Telcordia Family of Documents listed under FR-AMA-1. These documents specify standard mechanisms for generation and transmission of CDRs. You can specify one of four AMA flags to apply to all IAX connections:

amaflags=default|omit|billing|documentation

You can minimize the amount of authorization debugging by disabling it with authdebug=no. Authorization debugging is enabled by default if not explicitly disabled:

authdebug=no

To minimize the danger of stalling when a host is unreachable, you can set autokill to yes to specify that any new connection should be torn down if an ACK is not received within 2,000 ms. (This is obviously not advised for hosts with high latency.) Alternatively, you can replace yes with the number of milliseconds to wait before considering a peer unreachable. autokill configures the wait for all IAX2 peers, but you can configure it differently for individual peers with the use of the qualify command:

autokill=1500

bandwidth is a shortcut that may help you get around using disallow=all and multiple allow statements to specify which codecs to use. The valid options are:

These optional parameters allow you to control the IP interface and port on which you wish to accept IAX connections. If omitted, the port will be set to 4569, and all IP addresses in your Asterisk system will accept incoming IAX connections. If multiple bind addresses are configured, only the defined interface will accept IAX connections.^[160] The address 0.0.0.0 tells Asterisk to listen on all interfaces:

bindport=4569
bindaddr=192.168.0.1

The codecpriority option controls which end of an inbound call leg will have priority over the negotiation of codecs. If set in the [general] section, the selected options will be inherited by all user entries in the channel configuration file; however, they can be defined in the individual user entries for more granular control. If set in both the [general] and user sections, the user entry will override the entry that is configured in the [general] section. If this parameter is not configured, the value defaults to host.

Valid options include:

If an incorrect password is received on an IAX channel, this will delay the sending of the REGREQ or AUTHREP reject messages, which will help to secure against brute-force password attacks. The delay time is 1,000 ms:

delayreject=yes|no

Since Asterisk attempts to bridge channels (endpoints) directly together, the endpoints are normally allowed to perform jitter buffering themselves. However, if the endpoints have a poor jitter buffer implementation, you may wish to force Asterisk to perform jitter buffering no matter what. You can force jitter buffering to be performed with forcejitterbuffer=yes:

forcejitterbuffer=yes

The iaxthreads setting specifies the number of IAX helper threads that are created on startup to handle IAX communications.

The iaxmaxthreads setting specifies the maximum number of additional IAX helper threads that may be spawned to handle higher IAX traffic levels:

iaxthreads=10
iaxmaxthreads=100

Jitter refers to the varying latency between packets. When packets are sent from an end device, they are sent at a constant rate with very little latency variation. However, as the packets traverse the Internet, the latency between the packets may become varied; thus, they may arrive at the destination at different times, and possibly even out of order.

The jitter buffer is, in a sense, a staging area where the packets can be reordered and delivered in a regulated stream. Without a jitter buffer, the user may perceive anomalies in the stream, experienced as static, strange sound effects, garbled words, or, in severe cases, missed words or syllables.

The jitter buffer affects only data received from the far end. Any data you transmit will not be affected by your jitter buffer, as the far end will be responsible for the de-jittering of its incoming connections.

The jitter buffer is enabled with the use of jitterbuffer=yes:

jitterbuffer=yes|no

This sets the language flag to whatever you define. The global default language is English. The language that is set is sent by the channel as an information element. It is also used by applications such as SayNumber() that have different files for different languages. Keep in mind that languages other than English are not explicitly installed on the system, and it is up to you to configure the system to ensure that the language you specify is handled properly:

language=en

If mailboxdetail is set to yes, the new/old message count is sent to the user, instead of a simple statement of whether new and old messages exist. mailboxdetail can also be set on a per-peer basis:

mailboxdetail=yes

This parameter is used to set the maximum size of the jitter buffer, in milliseconds. Be sure not to set maxjitterbuffer too high, or you will needlessly increase your latency:

maxjitterbuffer=500

The maximum number of interpolation frames the jitter buffer should return in a row. Since some clients do not send CNG/DTX frames to indicate silence, the jitter buffer will assume silence has begun after returning this many interpolations. This prevents interpolating throughout a long silence:

maxjitterinterps=10

Specifies the maximum and minimum time intervals for registration expiration, in seconds:

maxregexpire=180
minregexpire=60

This option specifies a preference for which music-on-hold class this channel should listen to when put on hold if the music class has not been set on the channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer channel putting this all on hold did not suggest a music class.

If this option is set to passthrough, then the hold message will always be passed through as signaling instead of generating hold music locally.

This option may be specified globally, or on a per-user or per-peer basis:

mohinterpret=default

This option specifies which music-on-hold class (as defined in musiconhold.conf) to suggest to the peer channel when this channel places the peer on hold. It may be specified globally or on a per-user or per-peer basis:

mohsuggest=default

If set, Asterisk will disable the calculation of UDP checksums and no longer check UDP checksums on systems supporting this feature:

nochecksums=yes

By specifying the context that contains the actions to perform, you can configure Asterisk to perform a number of actions when a peer registers to your server. This option works in conjunction with regexten, which specifies the extension to execute. If no regexten is configured, the peer name is used as the extension. Asterisk will dynamically create and destroy a NoOp at priority 1 for the extension. All actions to be performed upon registration should start at priority 2. More than one regexten may be supplied, if separated by an &. regcontext can be set on a per-peer basis or globally:

regcontext=registered-phones

The regexten option is used in conjunction with regcontext to specify the extension to be executed within the configured context. If regexten is not explicitly configured, the peer name is used as the extension to match:

regexten=myphone

The resynchronize threshold is used to resynchronize the jitter buffer if a significant change is detected over a few frames, assuming that the change was caused by a timestamp mixup. The resynchronization threshold is defined as the measured jitter plus the resyncthreshold value, defined in milliseconds:

resyncthreshold=1000

This specifies whether or not Asterisk should auto-expire friends created on the fly on the same schedule as if they had just registered. If set to yes, when the registration expires, the friend will vanish from the configuration until requested again. If set to an integer, friends expire within that number of seconds instead of the normal registration interval:

rtautoclear=yes|no|seconds

If rtcachefriends is turned on, Asterisk will cache friends that come from the realtime engine, just as if they had come from iax.conf. This often helps with items such as message-waiting indications on realtime peers:

rtcachefriends=yes|no

If rtignoreregexpire is set to yes, and a realtime peer’s registration has expired (based on its registration interval), then Asterisk will continue to use the IP address and port stored in the database:

rtignoreregexpire=yes|no

If set to yes Asterisk will update the IP address, origination port, and registration period of a peer upon registration. Defaults to yes:

rtupdate=yes|no

Asterisk can set the Type of Service (TOS) bits in the IP header to help improve performance on routers that respect TOS bits in their routing calculations. The following values are valid: CS0, CS1, CS2, CS3, CS4, CS5, CS6, CS7, AF11, AF12, AF13, AF21, AF22, AF23, AF31, AF32, AF33, AF41, AF42, AF43, and ef (expedited forwarding). You may also use a numeric value for the TOS bits.

For more information, see the doc/ip-tos.txt file in the Asterisk source directory.

IAX2 trunking enables Asterisk to send media (as mini-frames) from multiple channels using a single header. The reduction in overhead makes the IAX2 protocol more efficient when sending multiple streams to the same endpoint (usually another Asterisk server):

trunk=yes|no

trunkfreq is used to control how frequently you send trunk messages, in milliseconds. Trunk messages are sent in conjunction with the trunk=yes command:

trunkfreq=20

Specifies whether or not Asterisk should send timestamps for the individual sub-frames within trunk frames. There is a small bandwidth penalty for sending these timestamps (less than 1 kbps/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 it does not also need to have it enabled:

trunktimestamps=yes|no