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Setting up a SIP Phone

This section describes the steps necessary to set up a SIP phone in ASL3. The setup procedure has changed due to the deprecation of chan_sip. Users are now required to use chan_pjsip. chan_pjsip brings new features to the SIP stack, and is the supported SIP channel for the future.

Update modules.conf

chan_pjsip requires a number of modules to be loaded. You should start by editing /etc/asterisk/modules.conf and add the following to the [modules] stanza (generally near the bottom of the file just above the [global] stanza):

;
; modules for pjsip
;
noload = app_voicemail.so
load = bridge_builtin_features.so
load = bridge_builtin_interval_features.so
load = bridge_holding.so
load = bridge_native_rtp.so
load = bridge_simple.so
load = bridge_softmix.so
load = chan_bridge_media.so
load = chan_pjsip.so
load = func_pjsip_endpoint.so
load = func_sorcery.so
load = func_devstate.so
load = res_pjproject.so
load = res_pjsip_acl.so
load = res_pjsip_authenticator_digest.so
load = res_pjsip_caller_id.so
load = res_pjsip_dialog_info_body_generator.so
load = res_pjsip_diversion.so
load = res_pjsip_dtmf_info.so
load = res_pjsip_endpoint_identifier_anonymous.so
load = res_pjsip_endpoint_identifier_ip.so
load = res_pjsip_endpoint_identifier_user.so
load = res_pjsip_exten_state.so
load = res_pjsip_header_funcs.so
load = res_pjsip_logger.so
load = res_pjsip_messaging.so
load = res_pjsip_mwi_body_generator.so
load = res_pjsip_mwi.so
load = res_pjsip_nat.so
load = res_pjsip_notify.so
load = res_pjsip_one_touch_record_info.so
load = res_pjsip_outbound_authenticator_digest.so
load = res_pjsip_outbound_publish.so
load = res_pjsip_outbound_registration.so
load = res_pjsip_path.so
load = res_pjsip_pidf_body_generator.so
load = res_pjsip_pidf_digium_body_supplement.so
load = res_pjsip_pidf_eyebeam_body_supplement.so
load = res_pjsip_publish_asterisk.so
load = res_pjsip_pubsub.so
load = res_pjsip_refer.so
load = res_pjsip_registrar.so
load = res_pjsip_rfc3326.so
load = res_pjsip_sdp_rtp.so
load = res_pjsip_send_to_voicemail.so
load = res_pjsip_session.so
load = res_pjsip.so
noload = res_pjsip_t38.so
noload = res_pjsip_transport_websocket.so
load = res_pjsip_xpidf_body_generator.so
load = res_rtp_asterisk.so
load = res_sorcery_astdb.so
load = res_sorcery_config.so
load = res_sorcery_memory.so
load = res_sorcery_realtime.so

Update extensions.conf

The next step is to update extension.conf. This creates a dial plan that controls access to your node by SIP phones. Edit /etc/asterisk/extensions.conf and add the following to the bottom of the file. This example will set up extension 1001 as your local phone and the dialing plan for your node.

[sip-phones]
exten => 1001,1,Dial(PJSIP/${EXTEN},60,rT)

exten => ${NODE},1,Ringing
exten => ${NODE},n,Answer(3000)
exten => ${NODE},n,Set(NODENUM=${CALLERID(number)})
exten => ${NODE},n,Playback(extension)
exten => ${NODE},n,SayDigits(${NODENUM})
exten => ${NODE},n,Playback(rpt/connected)
exten => ${NODE},n,Playback(rpt/node)
exten => ${NODE},n,SayDigits(${EXTEN})
exten => ${NODE},n,rpt(${EXTEN}|P)
exten => ${NODE},n,Hangup

The above dial plan will allow your SIP phone to dial your node number. In this example, one should use *99 to enable PTT and # to disable PTT from the SIP phone. VOX is also an option for phone connections. When using VOX, one may want to mute the mic on the phone when not transmitting and un-mute to transmit. VOX is enabled by adding a v to the rpt line like this:

exten => ${NODE},n,rpt(${EXTEN}|Pv)

This dial plan announces the extension number and the connecting node number. The lines with SayDigits and Playback can be removed if you do not want the announcement.

The ${NODE} variable is defined at the top of extensions.conf and should be your assigned node number. See the following:

[globals]
HOMENPA = 999 ; change this to your Area Code
NODE = 1999   ; change this to your node number

NOTE: If you have multiple node numbers configured, you may need to adjust the ${NODE} variable to the appropriate node number, or replace the ${NODE} variable in your extensions.conf stanza with the specific node number desired.

The entry 1001 in extensions.conf is the extension number for your SIP phone. If you want to use a different extension number, change 1001 to your desired extension number.

NOTE: If you attempt to copy extensions.conf from a previous release of AllStarLink, it will fail. ASL3 requires comma delimiters instead of pipe symbols for standard Asterisk functions. The rpt function continues to use a pipe delimiter. This is subject to change in a future release.

Update pjsip.conf

You are now ready to configure pjsip. Edit /etc/asterisk/pjsip.conf. Don’t get overwhelmed by all of the text in the sample configuration. pjsip supports several configurations. This document focuses on the necessary entries to get a SIP phone operational.

Scroll down to the Basic UDP transport section. It should be updated to look like the following:

; Basic UDP transport
;
[transport-udp]
type=transport
protocol=udp    ;udp,tcp,tls,ws,wss,flow
bind=0.0.0.0:5060

NOTE: After the bind port address (0.0.0.0) is the incoming port number (5060). This UDP port must NOT be blocked by your network firewall. If you are using a Pi Applicance, you will likely need to add the port to the built-in firewall.

Scroll down to the section titled Endpoint Configured For Use With A Sip Phone. For each extension, you will need to enter three sections. Here is an example for extension 1001:

[1001]
type=endpoint
transport=transport-udp
context=sip-phones
disallow=all
allow=ulaw
allow=alaw
allow=gsm
auth=1001
aors=1001
callerid="My CallerID"
;
; A few more transports to pick from, and some related options below them.
;
;transport=transport-tls
;media_encryption=sdes
;transport=transport-udp-ipv6
;transport=transport-udp-nat
;direct_media=no
;
; MWI related options

;aggregate_mwi=yes
;mailboxes=6001@default,7001@default
;mwi_from_user=6001
;
; Extension and Device state options
;
;device_state_busy_at=1
;allow_subscribe=yes
;sub_min_expiry=30
;
; STIR/SHAKEN support.
;
;stir_shaken=no
;stir_shaken_profile=my_profile

[1001]
type=auth
auth_type=userpass
password=1001
username=1001

[1001]
type=aor
max_contacts=4
;contact=sip:6001@192.0.2.1:5060

If you want to use a different extension number, you will need to change [1001] to your desired number. The item callerid="My CallerID" should also be updated for your preferred caller identifier.

The items password=1001 and username=1001 should be updated to match the username and password you want for your extension number. You should use a complex password. Do not use the extension number. You can use the password generator located here to generate a complex password.

These three sections can be replicated for each extension that you want to add to the system.

Restart Asterisk

After completing these changes, you must restart Asterisk or reboot your system.

sudo systemctl restart asterisk

Example Config with Multiple SIP Phones

As noted above you can have more SIP phones. Here is an example of a simple dial plan that has multiple extensions:

[sip-phones]
; Extension 210 - Line 1
; Extension 211 - Line 2
; Extension 212 - Garage
; Extension 213 - Cordless ATA
; Extension 1000 - Voice Mail

exten => 210,1,Dial(PJSIP/210,60,rT)
exten => 211,1,Dial(PJSIP/211,60,rT)
exten => 212,1,Dial(PJSIP/212,60,rT)
exten => 213,1,Dial(PJSIP/213,60,rT)
exten => 1000,1,VoiceMailMain(210)
exten => 1000,2,Hangup

; Allow SIP calls to local nodes
exten => 1998,1,rpt(1998|P)
exten => 1999,1,rpt(1999|P)

This example includes voicemail. To use voice mail, you will need to change noload = app_voicemail.soto load = app_voicemail.so.

Troubleshooting

If you have trouble connecting your SIP phone, start the Asterisk CLI with asterisk -rvvv. Enter the command pjsip set logger on. This will show the communications between the SIP phone and Asterisk. It will also show the actions in your dial plan.

Security

If you will be exposing your system to the outside world, you should consider using fail2ban to add additional protection for the system.