We'll provide two examples, but you should look at the documentation for the channel driver or feature to better understand how to configure media in that context. In the general section of voicemail. We set the option "format" to a string of file format names. Asterisk supports 8, 16, and 32kHz Speex. Use of the 32kHz Speex mode is, like the other modes, controlled in the respective channel driver's configuration file, e.
The complete list of supported sampling rates and file format is found in the expansion link below:. Users can create bit Signed Linear files of varying sampling rates from WAV files using the sox command-line audio utility.
The resulting output. Video transcoding or image transcoding is not currently supported. It should be noted that direct media is not supported for video calls.
Evaluate Confluence today. Asterisk Project Home Operation. See the auth realm description for details. Endpoints and AORs can be identified in multiple ways.
This option is a comma separated list of methods the endpoint can be identified. This option controls both how an endpoint is matched for incoming traffic and also how an AOR is determined if a registration occurs. You must list at least one method that also matches for AORs or the registration will fail.
This method of identification has some security considerations because an Authentication header is not present on the first message of a dialog when digest authentication is used. The client can't generate it until the server sends the challenge in a response. Since Asterisk normally sends a security event when an incoming request can't be matched to an endpoint, using this method requires that the security event be deferred until a request is received with the Authentication header and only generated if the username doesn't result in a match.
This may result in a delay before an attack is recognized. When a redirect is received from an endpoint there are multiple ways it can be handled. If this option is set to user the user portion of the redirect target is treated as an extension within the dialplan and dialed using a Local channel.
More than one mailbox can be specified with a comma-delimited string. On inbound SIP messages from this endpoint, the Contact header or an appropriate Record-Route header will be changed to have the source IP address and port.
This option does not affect outbound messages sent to this endpoint. This option helps servers communicate with endpoints that are behind NATs. When enabled, immediately send Ringing or Progress response messages to the caller if the connected line information is updated before the call is answered. This can send a Ringing response before the call has even reached the far end.
The caller can start hearing ringback before the far end even gets the call. Many phones tend to grab the first connected line information and refuse to update the display if it changes. The first information is not likely to be correct if the call goes to an endpoint not under the control of this Asterisk box. When disabled, a connected line update must wait for another reason to send a message with the connected line information to the caller before the call is answered.
You can trigger the sending of the information by using an appropriate dialplan application such as Ringing. This will force the endpoint to use the specified transport configuration to send SIP messages. Not specifying a transport will select the first configured transport in pjsip.
Transport configuration is not affected by reloads. In order to change transports, a full Asterisk restart is required. This option determines whether Asterisk will accept identification from the endpoint from headers such as P-Asserted-Identity or Remote-Party-ID header.
This option applies both to calls originating from the endpoint and calls originating from Asterisk. If no , the configured Caller-ID from pjsip.
If no , private Caller-ID information will not be forwarded to the endpoint. When set to "yes" and an endpoint negotiates g. Since this essentially replaces the underlying 'g' codec with 'gaal2' then 'gaal2' needs to be specified in the endpoint's allowed codec list. Can be set to a comma separated list of numbers or ranges between the values of maximum of 64 groups.
If set to yes T. This option can be set to override the maximum datagram of a remote endpoint for broken endpoints. Setting the value to zero disables the timeout. When an INFO request for one-touch recording arrives with a Record header set to "on", this feature will be enabled for the channel. The feature designated here can be any built-in or dynamic feature defined in features.
When an INFO request for one-touch recording arrives with a Record header set to "off", this feature will be enabled for the channel. If enabled, Asterisk will generate an X. When a new channel is created using the endpoint set the specified variable s on that channel. If specified, any channel created for this endpoint will automatically have this accountcode set on it. At the specified interval, Asterisk will send an RTP comfort noise frame.
This may be useful for situations where Asterisk is behind a NAT or firewall and must keep a hole open in order to allow for media to arrive at Asterisk. This option configures the number of seconds without RTP while off hold before considering a channel as dead.
When the number of seconds is reached the underlying channel is hung up. By default this option is set to 0, which means do not check. This option configures the number of seconds without RTP while on hold before considering a channel as dead. This matches sections configured in acl. The value is defined as a list of comma-delimited section names. The value is a comma-delimited list of IP addresses.
IP addresses may have a subnet mask appended. The subnet mask may be written in either CIDR or dotted-decimal notation. When set to "yes" the codec in use for sending will be allowed to differ from that of the received one. PJSIP will not automatically switch the sending one to the receiving one. With this option enabled, Asterisk will attempt to negotiate the use of the "rtcp-mux" attribute on all media streams.
Many people will tell you that G. This is not exactly true, as companding is considered a form of compression. What is true is that G. This codec has been around for some time it used to be G. The most common rates are 16 Kbps, 24 Kbps, and 32 Kbps. As of this writing, Asterisk supports only the ADPCM rate, which is far and away the most popular rate for this codec. This is possible because rather than sending the result of the quantization measurement, it sends only enough information to describe the difference between the current sample and the previous one.
Considering how little bandwidth it uses, G. To achieve its impressive compression ratio, this codec requires an equally impressive amount of effort from the CPU. In an Asterisk system, the use of heavily compressed codecs will quickly bog down the CPU.
This codec does not come encumbered with a licensing requirement the way that G. The sound quality is generally considered to be of a lesser grade than that produced by G. GSM operates at 13 Kbps. The Internet Low Bitrate Codec iLBC provides an attractive mix of low bandwidth usage and quality, and it is especially well suited to sustaining reasonable quality on lossy network links. Naturally, Asterisk supports it and support elsewhere is growing , but it is not as popular as the ITU codecs, and thus may not be compatible with common IP telephones and commercial VoIP systems.
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