Asterisk Configuration files Asterisk.conf defines the locations for the configuration files, the spool directory, and the modules, as well as a location to write log files to. [directories] astetcdir => /etc/asterisk ; location where Asterisk configuration files are stored astmoddir => /usr/lib/asterisk/modules ; location where Asterisk modules are stored astvarlibdir => /var/lib/asterisk ; location where Asterisk libraries are stored astagidir => /var/lib/asterisk/agi-bin ;location where agi scripts are stored astspooldir => /var/spool/asterisk ; location where all records are stored astrundir => /var/run ; location where Asterisk will store its PID file astlogdir => /var/log/asterisk ; location where Asterisk logs are stored

Manager.conf The Asterisk Manager interface is an API which external programs can use to communicate with, and control, Asterisk. [general] enabled = no port = 5038 bindaddr = 0.0.0.0 [magma] secret = welcome deny=0.0.0.0/0.0.0.0 permit= 192.168.1.0/255.255.255.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user

cdr_custom.conf Asterisk will generate a CDR for every finished call by default; this file defines exactly what gets logged in this CDR line. [mappings] Master.csv => "${CDR(clid)}","${CDR(src)}","${CDR(dst)}","${CDR(dcontext)}","${CDR(channel)}","${CDR(dstchannel)}","${CDR(lastapp)}","${CDR(lastdata)}","${CDR(start)}","${CDR(answer)}","${CDR(end)}","${CDR(duration)}","${CDR(billsec)}","${CDR(disposition)}","${CDR(amaflags)}","${CDR(accountcode)}","${CDR(uniqueid)}","${CDR(userfield)}”

rtp.conf The rtp.conf file controls the Real-time Transport Protocol (RTP) ports that Asterisk uses to generate and receive RTP traffic. [general] rtpstart=10000 rtpend=20000 rtpchecksums=no

Sip.conf [general} section Implements settings on all SIP clients Port The port Asterisk should listen for SIP incoming clients. The default is 5060 Bindaddr If the Asterisk machine has multiple IPs, this option tells Asterisk to listen on a particular IP. Asterisk listens on all interfaces and aliases by default Context The default context for SIP clients

Sip.conf Type of Service (ToS) Sets ToS bits in an IP header to improve performance on routers that respects ToS bits The options are low delay, throughput, reliability, mincost, none Allow and disallow Allows and disallows specific codecs. Codec negotiation is attempted in the order in which codecs are defined. The best practice is to disallow all and then allow a codec explicitly For example: disallow=all allow=g729 allow=alaw allow=ulaw

Sip.conf Maxexpirey The maximum amount of time until a peer’s registration expires Defaultexpirey This sets the default SIP registration expiration time, in seconds, for incoming and outgoing registrations Register Register this Asterisk with another host

Sip.conf Client Options Type The type option sets the connection class for the client. The options are: peer: A device which receives calls from an Asterisk server user: A device that makes calls through an Asterisk server friend: A device that can both send and receive calls through an Asterisk server Secret Set a password for the client Host A client’s IP address. Can also be set to dynamic if a client comes from any IP

Sip.conf Defaultip Asterisk will send a call on this IP if a host is set to dynamic and the SIP client is not registered yet Username A client’s username Context The context to start in extensions.conf Canreinvite Connects end-points directly Nat Can be set to yes, no, or never. If set to yes, Asterisk ignores the IP address in the SIP and SDP headers and responds to the address and port in the IP header. The never option is for devices that cannot handle rport in the SIP header, such as the Uniden UIP200.

Sip.conf Qualify Sets qualify to yes, no, or a time in milliseconds. If you set qualify=yes, NOTIFY messages will be sent periodically to the remote peers to determine whether they are available and what the latency between replies is. A peer is determined unreachable if no reply is received within 2,000 ms (to change this default, set qualify to the number of milliseconds to wait for the reply). Use this option in conjunction with nat=yes to keep the path through the NAT device alive.

Iax.conf [general] section Bandwidth Instead of allowing or disallowing codecs, use low or high bandwidth options to automatically select the low bandwidth consumption codec or high bandwidth consumption codec jitterbuffer Turn the jitterbuffer on or off. The jitterbuffer is used to maximize audio quality by balancing latency against the number of dropped packets dropcount Sets the maximum number of packets per memory size to be dropped in order to reduce latency maxjitterbuffer Sets the maximum size of the jitterbuffer

Iax.conf maxexcessjitterbuffer: Sets the maximum excess jitterbuffer, which, if exceeded, causes the jitterbuffer to slowly shrink in order to improve latency.