# Asterisk pjsip_wizard.conf: Endpoints and Trunks in a Few Lines Source: https://srvscripts.com/guides/asterisk-pjsip-wizard/ Updated: 2026-10-07 Publisher: srvScripts (https://srvscripts.com/) **Short answer:** `pjsip_wizard.conf` (module `res_pjsip_config_wizard`) lets one `type = wizard` section create the endpoint, AOR, auth, identify and registration objects that plain `pjsip.conf` needs separately. Set `accepts_registrations` and `accepts_auth` for phones, `remote_hosts` with `sends_auth` and `sends_registrations` for a registering trunk, or `remote_hosts` alone for an IP-authenticated trunk, and pass any other option with prefixes such as `endpoint/` and `aor/`. It is for plain Asterisk: on FreePBX, define extensions and trunks in the GUI. We ran the configuration and commands below on our lab server (Debian 12, Asterisk 22.11) on 7 October 2026, on a separate test Asterisk instance bound to 127.0.0.1 with documentation IP addresses for the trunks, so nothing registered to a real provider. We also checked the FreePBX 17.0.33 instance on the same server read-only. ## What the wizard generates A normal PJSIP phone needs three sections in `pjsip.conf` (endpoint, aor, auth) and a registering trunk needs five or six. The wizard builds them from one section and names them predictably. From the Asterisk documentation, and as we saw on our test instance: | Object | Name | Created when | | --- | --- | --- | | endpoint | same as the wizard section | always | | aor | same as the wizard section | always; one contact per remote_hosts entry | | inbound auth | -iauth | accepts_auth = yes | | outbound auth | -oauth | sends_auth = yes | | registration | -reg-0, -reg-1… | sends_registrations = yes, one per remote host | | identify | -identify | remote_hosts is set (not with accepts_registrations) | | phoneprov | -phoneprov | has_phoneprov = yes | Every generated object carries an `@pjsip_wizard` attribute with the wizard name, visible in `pjsip show endpoint`, `pjsip show aor` and similar commands. That is how you tell wizard objects from hand-written ones. Options that do not have a wizard keyword are passed through with a prefix: `endpoint/`, `aor/`, `inbound_auth/`, `outbound_auth/`, `registration/`, `identify/` and `phoneprov/`. So `endpoint/context = from-phones` becomes `context = from-phones` on the endpoint. `config show help res_pjsip_config_wizard wizard` on Asterisk 22.11 lists the wizard keywords: - `type`, `transport`, `remote_hosts`, `outbound_proxy` - `sends_auth`, `accepts_auth`, `sends_registrations`, `accepts_registrations`, `sends_line_with_registrations` - `server_uri_pattern`, `client_uri_pattern`, `contact_pattern` - `has_phoneprov`, `has_hint`, `hint_context`, `hint_exten`, `hint_application` ## Before you start: transport in pjsip.conf The wizard does not create transports, `system` or `global` sections. Those stay in `pjsip.conf`, and the module must be loaded (`asterisk -rx "module show like wizard"` should list `res_pjsip_config_wizard.so` as Running). ``` ; /etc/asterisk/pjsip.conf [transport-udp] type = transport protocol = udp bind = 0.0.0.0:5060 ``` If the server is behind NAT, add `external_media_address`, `external_signaling_address` and `local_net` to the transport; our [PJSIP NAT generator](/tools/pjsip-nat-generator/) writes them for you. ## Phones: one template, three lines per extension Templates (a section name followed by `(!)`) hold the shared settings; each phone inherits with `[name](template)`. This is the file we loaded on our test instance, with placeholder passwords: ``` ; /etc/asterisk/pjsip_wizard.conf [phone-defaults](!) type = wizard transport = transport-udp accepts_registrations = yes accepts_auth = yes endpoint/context = from-phones endpoint/disallow = all endpoint/allow = ulaw,alaw endpoint/direct_media = no aor/max_contacts = 1 aor/remove_existing = yes aor/qualify_frequency = 60 [1001](phone-defaults) inbound_auth/username = 1001 inbound_auth/password = USE-A-LONG-RANDOM-PASSWORD endpoint/callerid = "Bob" [1002](phone-defaults) inbound_auth/username = 1002 inbound_auth/password = USE-A-LONG-RANDOM-PASSWORD endpoint/callerid = "Alice" ``` - `accepts_registrations = yes`: the phone registers to Asterisk, so the AOR gets its contact from the REGISTER. No identify object is created. - `accepts_auth = yes`: creates `1001-iauth` from the `inbound_auth/` lines and sets it as the endpoint’s auth. - `aor/max_contacts = 1` with `remove_existing = yes`: a new registration replaces the old one instead of being rejected. - The context `from-phones` must exist in `extensions.conf`. Keep phones and trunks in different contexts so a call from a trunk can never reach your outbound routes. If you want Asterisk to create hints too, add `has_hint = yes`, `hint_context` and `hint_exten`; the Asterisk documentation shows this pattern. ## ITSP trunks: registration and IP authentication A trunk wizard uses `remote_hosts`. With registration and outbound authentication: ``` [trunk-defaults](!) type = wizard transport = transport-udp endpoint/context = from-itsp endpoint/disallow = all endpoint/allow = ulaw,alaw aor/qualify_frequency = 60 ; provider that wants you to register with a user name and password [itsp-reg](trunk-defaults) remote_hosts = 203.0.113.10 sends_auth = yes sends_registrations = yes outbound_auth/username = 4420000000 outbound_auth/password = PROVIDER-PASSWORD registration/expiration = 300 registration/retry_interval = 60 ; provider that authenticates you by IP address only [itsp-ip](trunk-defaults) remote_hosts = 203.0.113.20 sends_auth = no sends_registrations = no ``` - `sends_registrations = yes` builds `itsp-reg-reg-0` with `server_uri = sip:203.0.113.10` and `client_uri = sip:4420000000@203.0.113.10`, from the default patterns `sip:${REMOTE_HOST}` and `sip:${USERNAME}@${REMOTE_HOST}`. Change them with `server_uri_pattern` and `client_uri_pattern` if your provider wants a different domain. - `sends_auth = yes` creates `itsp-reg-oauth` and uses it for both the registration and outbound calls. - Both trunks get an identify object matching the remote host’s address (`203.0.113.10/32` and `203.0.113.20/32` on our test), so calls from the provider land on the right endpoint and context. - `remote_hosts` takes a comma-separated list, and a host can include a port (`sip.example.com:5080`). With several hosts you get one contact, one registration and one identify match per host. - If the provider sends calls from addresses you cannot list, `sends_line_with_registrations = yes` adds the `line` and `endpoint` parameters to the registration so inbound calls can be matched to it. For the full trunk picture (routes, caller ID, NAT), see [FreePBX PJSIP trunk setup](/guides/freepbx-pjsip-trunk-setup/); the provider side is the same on plain Asterisk. ## Load and reload the wizard ``` asterisk -rx "module reload res_pjsip_config_wizard.so" ``` On our test instance, adding a phone section and running that command printed `Module 'res_pjsip_config_wizard.so' reloaded successfully`, and the new endpoint appeared without a restart. A full `core reload` also re-reads it. Note there is no `pjsip reload` command; Asterisk 22.11 answered “No such command”. ## Check that it worked `pjsip show endpoints` lists everything the wizard built. From our test instance (no phones registered, trunks pointing at documentation addresses, so all show Unavailable): ``` Endpoint: 1001/1001 Unavailable 0 of inf InAuth: 1001-iauth/1001 Aor: 1001 1 Transport: transport-udp udp 0 0 127.0.0.1:5099 Endpoint: itsp-ip Unavailable 0 of inf Aor: itsp-ip 0 Contact: itsp-ip/sip:203.0.113.20 1e48d23f09 NonQual nan Transport: transport-udp udp 0 0 127.0.0.1:5099 Identify: itsp-ip-identify/itsp-ip Match: 203.0.113.20/32 Endpoint: itsp-reg Unavailable 0 of inf OutAuth: itsp-reg-oauth/4420000000 Aor: itsp-reg 0 Contact: itsp-reg/sip:203.0.113.10 aad5b6aade NonQual nan Transport: transport-udp udp 0 0 127.0.0.1:5099 Identify: itsp-reg-identify/itsp-reg Match: 203.0.113.10/32 ``` More checks: - `pjsip show registrations`: on a real server the trunk should show `Registered`. Ours showed `Rejected`, as expected, because the test transport was bound to 127.0.0.1 and could not reach the documentation address. - `pjsip show registration itsp-reg-reg-0`: confirms `server_uri`, `client_uri`, `expiration` and `outbound_auth`. - `pjsip show endpoint 1001`: the `@pjsip_wizard` line, `context`, `allow` and `callerid` should match what you set. - `pjsip show contacts`: after a phone registers, its contact appears with a status and round-trip time. - `pjsip show identifies`: one entry per trunk host. ## FreePBX and the wizard FreePBX writes its own PJSIP files (`pjsip.endpoint.conf`, `pjsip.aor.conf`, `pjsip.auth.conf`, `pjsip.registration.conf` and so on, included from `pjsip.conf`) from the database every time you apply configuration. On our FreePBX 17.0.33 lab, `res_pjsip_config_wizard.so` is loaded and `/etc/asterisk/pjsip_wizard.conf` exists, but it contains only comments. Do not add extensions or trunks there on a FreePBX system. FreePBX does not know about them: they do not show in the GUI, FreePBX’s dialplan does not treat them as extensions or trunks, and the documentation does not allow a wizard name to collide with an existing object, such as one FreePBX generated. Use Applications > Extensions and Connectivity > Trunks instead. The wizard is the right tool for plain Asterisk builds, such as one installed with our [Asterisk 22 on Debian guide](/guides/install-asterisk-22-debian/). ## Common problems - **Endpoint missing after reload:** generated objects must pass the same checks as hand-written ones, so an invalid option value can stop one from loading. Look in the Asterisk log for errors naming the wizard section. - **`accepts_registrations` together with `remote_hosts`:** the documentation says they are mutually exclusive. Use one or the other. - **Name collision:** a wizard section and a hand-written endpoint, AOR or auth with the same name cannot coexist. - **Inbound trunk calls rejected as unknown:** the provider sends from an address not in `remote_hosts`. Add the address, or pass extra `identify/match` values. - **Registration stays Rejected or Unregistered:** check credentials, the provider’s expected client URI and firewall rules; see [SIP error codes](/guides/sip-response-codes/) and [sngrep](/guides/troubleshoot-sip-sngrep/). **Official documentation:** [Asterisk: PJSIP Configuration Wizard](https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/PJSIP-Configuration-Wizard/) · [Asterisk: res_pjsip_config_wizard reference](https://docs.asterisk.org/Latest_API/API_Documentation/Module_Configuration/res_pjsip_config_wizard/) **Related:** [chan_sip to PJSIP Migration: sip_to_pjsip.py, Option Mapping, Tests](/guides/chan-sip-to-pjsip-migration/) · [FreePBX PJSIP Trunk Setup: Registration, IP Auth, Routes, NAT](/guides/freepbx-pjsip-trunk-setup/) · [Install Asterisk 22 LTS on Debian 13: Step-by-Step Guide](/guides/install-asterisk-22-debian/) · [PJSIP Endpoint Unreachable or Unavailable: Fix Qualify and NAT](/guides/pjsip-endpoint-unreachable-qualify/) · [PJSIP NAT Settings Generator: Asterisk, FreePBX, Issabel, VitalPBX, 3CX](/tools/pjsip-nat-generator/) **See also:** [FreePBX PJSIP Trunk Setup: Registration, IP Auth, Routes, NAT](/guides/freepbx-pjsip-trunk-setup/) · [Provision Yealink Phones on FreePBX 17 (Without Commercial EPM)](/guides/yealink-provisioning-freepbx/) · [Asterisk AMI Originate a Call with Python](/guides/asterisk-ami-originate-call-python/) ## Frequently asked questions ### What is pjsip_wizard.conf in Asterisk? A configuration file read by res_pjsip_config_wizard. Each type = wizard section generates the endpoint, AOR, auth, identify and registration objects you would otherwise write in pjsip.conf. ### How do I reload pjsip_wizard.conf? Run asterisk -rx “module reload res_pjsip_config_wizard.so”, or core reload. There is no pjsip reload command. ### Can I use pjsip_wizard.conf on FreePBX? The module is loaded on FreePBX 17, but you should not. FreePBX generates its own PJSIP files and will not know about wizard objects. Use the GUI. ### How do I set endpoint options that the wizard does not have? Prefix them: endpoint/context, endpoint/allow, aor/max_contacts, registration/expiration and so on are passed straight to the generated object. ### What is the difference between sends_auth and accepts_auth? sends_auth creates outbound credentials that Asterisk uses towards a provider. accepts_auth creates inbound credentials that phones must use towards Asterisk.