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INSTRUCTIONS.md

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What is it?

SVD is a simple sip client using the FXS ports on the infineon danube. It uses sofia-sip to manage the sip part and a custom written library (libab) to manage the ports. You can define multiple sip accounts, decide which FXS ports to use for outgoing calls and which ports to ring for incoming calls.

Placing a call

To place a call simply dial a number once you hear the dialtone. The call will be placed after 4 seconds you keyed in the last digit or after you dial # (the # won't be part of the dialled number). To determine the account to use to place the call, first the dialplan will be checked, and if no entry matches, the call will be placed on the sip account with the lowest priority. See below for instructions about the dialplan and the configuration of the accounts.

Configuration

This program uses uci for configuration, there's a web interface available in the package luci-svd but you can also manually edit the /etc/config/svd file. Various sections can be defined:

  • config main

general parameters

  • config account name

parameters for a sip account, you can define as many sections as accounts you want to use, each with a different name

  • config dialplan

dialplan entries (see below), as many as you want

  • config channel number

overrides defaults parameters for one channel (number can be 1 or 2).

  • config codec name

overrides default codec parameter. Name must be one of:

G722, PCMA, G729, G729E, G723, iLBC, G726-16, G726-32, G726-40, telephone-event

repeat this section for each codec you want to modify

options in config main

  • option log_level n

optional - changes the logging level between 0 (quiet) and 9 (verbose)

  • option rtp_port_first n

mandatory - number of the first port to use for rtp

  • option rtp_port_last n

mandatory - number of the last port to use for rtp

  • option sip_tos n

mandatory -tos for sip traffic

  • option rtp_tos n

mandatory - tos for rtp traffic

  • option led name

optional - name of the led to use as the main voip led. If configured the led will turn on when there's at least one account successfully registered.

example:

	config main
	    option log_level 5
	    option rtp_port_first 5500
	    option rtp_port_last 5600
	    option sip_tos 0x10
	    option rtp_tos 0x10
            option led "voice"

options in config account

  • option user "username"

mandatory, username of the account

  • option domain "domain"

mandatory, domain of the account

  • option password "password"

mandatory, password of the account

  • option display "display_name"

optional, sip display name. It will be used as is, if quotes are needed you'll have to write them yourself, e.g. '"display name"'

  • option registrar "registrar"

optional, host name of the registrar, if not defined "domain" will be used

  • option auth_name "auth_name"

optional, username for authentication, if not defined "user" will be used

  • option outbound_proxy "proxy"

optional, proxy to be used for outgoing calls. If not defined no outbound proxy will be used.

  • option codecs "codec1 codec2 codec3 ..."

optional, list of codecs to use ordered by priority. If not defined, all codecs will be used in this order: PCMA, G729, G729E, G723, iLBC, G726-16, G726-32, G726-40

  • option ring 'off on'

optional, list of booleans. Decides which channel will ring for incoming calls to this account. If not defined, all channels will ring.

  • option priority '0 1'

optional, list of integers (one integer for each channel). When a number is dialled on a channel, first the dialplan will be tried to decide the account to use. If no dialplan entry matches, the lowest priority correctly registered account will be used. An account with 0 priority will not be used.

  • option dtmf type

optional, how to send dtmf on this account. type can be "rfc2883", "info" or "off". if not defined, "rfc2833" will be used.

  • option disabled on

optional, use it to disable an account without removing it from the configuration.

  • option user_agent "My user agent"

optional user agent to use instead of the default (some sip registrars only allow registration coming with a specific user agent, e.g. Jazztel in Spain).

examples:

minimal configuration

       config account test1
           option user "testuser"
           option domain "sip.test.domain"
           option password "testpassword"

this account will ring every channel and it will be used by every channel for outgoing calls.

more complete configuration

       config account test2
           option user "testuser2"
           option domain "sip2.test.domain"
           option auth_name "authuser"
           option display '"my display name"'
           option registrar "proxy.test.domain"
           option password "testpassword"
           option outbound_proxy "outproxy.test.domain"
           option codecs "g729 pcma"
           option ring 'off on'
           option priority '0 7'
           option dtmf info

this account will only ring the second channel and it will only be used for outgoing calls by the second channel, provided there's no other channel with a priority lower than 7.

options in config dialplan

  • option prefix "1234"

mandatory, prefix of the dialled number that matches this entry

  • option replace "abcd"

optional, the matched prefix will be replaced by this string

  • option remove_prefix on

optional, if set to true, the prefix will be removed before calling. It will have no effect if "option replace" is present.

  • option account name

mandatory, name of the account to use

Example:

    config dialplan
      option prefix "00"
      option account "test1"

   config dialplan
      option prefix "777"
      option replace "an_user"
      option account "test2"

when a number starting with 00 is dialled, account "test1" will be used

(e.g. 001234 will call 001234@sip.test.domain)

when a number starting with 777 is dialled, account "test2" will be used, replacing "777" with "an_user"

(e.g. 777 will call an_user@sip2.test.domain)

options in config channel

  • option enc_db

<integer>

default: 0

  • option dec_db

<integer>

default: 0

  • option vad

<string>

can be one of "off", "on", "g711", "cng_only", "sc_only" default: "off"

  • option hpf

<boolean>

high pass filter, on or off default: "off"

  • option wlec_type

<string>

can be one of "off", "ne", "nfe" default: "ne"

  • option wlec_nlp

<boolean>

can be "on" or "off" default: "off"

  • option wlec_ne_nb

<integer>

default: 4

  • option wlec_fe_nb

<integer>

default: 4

  • option cid

<string>

sets the caller id standard, can be one of:

"off", "telcordia", "etsi_fsk", "etsi_dtmf", "sin", "ntt", "kpn_dtmf", "kpn_dtmf_fsk"

defaults to "etsi_fsk", "off" disables caller id transmission.

  • option led

<string>

name of the led used by this channel. If configured, this led will turn on when the phone is off-hook, it will blink slowly if is there a call ongoing and it will blink fast if the phone is ringing.

Example:

    config channel 1
       option cid telcordia
       option led "fxs2"
    config channel 2
       option cid ntt
       option led "fxs1"

options in config codec

  • option packet_size

<string>

can be 2.5, 5, 5.5, 10, 11, 20, 30, 40, 50, 60

  • option payload

<integer>

payload type to use for rtp

  • option bitpack

<string>

can be rtp or aal2

  • option jb_type

<string>

jitter buffer, can be fixed or adaptive

  • option scaling

<string>

actually a float

  • option local_adaptation

<boolean>

can be on or off

  • option jb_init_sz

<integer>

initial size for the jitter buffer

  • option jb_min_sz

<integer>

minimum size for the jitter buffer

  • option jb_max_sz

<integer>

maximum size for the jitter buffer