# Asterisk PJSIP Configuration (v20-23)

Connect a plain Asterisk installation to DIDWW by using `chan_pjsip`. This
guide covers inbound calls through a Static Endpoint or Dynamic Registration
trunk and outbound calls through a DIDWW outbound trunk.

For general information about the channel driver and its configuration model,
see the official [Configuring
res\_pjsip](https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/)
documentation.

This configuration applies to **Asterisk 20, 21, 22, and 23 versions** and uses the configuration schema included with
stock Asterisk.
It does not require third-party patches.

Note

For Asterisk installations managed through FreePBX, use the
[FreePBX](../freepbx/index.html) guide. For an older installation that still has
`chan_sip` installed, see the [Legacy Asterisk Configuration](legacy-versions.html) guide.

---

## 1. Configure inbound voice trunk

Configure an Inbound SIP Trunk in the DIDWW User Panel to send incoming calls
from your DIDWW numbers to Asterisk. Select the delivery method that matches
your Asterisk deployment.

### Before you begin

- An active DIDWW account is required. [Sign in to DIDWW](https://my.didww.com/users/sign_in) or [create a DIDWW account](https://my.didww.com/users/sign_up#/users/sign_up).
- At least one active **DID number** with capacity to receive incoming calls is
  required. If you do not have one, see
  [How to buy numbers](../../phone-numbers/buy-numbers/how-to-buy.html).

### Step 1: Create New Inbound SIP trunk

1. In the [DIDWW User Panel](https://my.didww.com/#/trunks), go to
   **Voice > Inbound Trunks**.
2. Click **Create New > SIP Trunk**.

[![Creating a new inbound SIP trunk](../../_images/create-inbound-sip-trunk.png)](../../_images/create-inbound-sip-trunk.png)


Creating a new inbound SIP trunk.

### Step 2: Configure general SIP trunk settings

In the Create Inbound SIP Trunk form, enter the settings for the selected
trunk type.

Static Endpoint

1. Enter a descriptive **Name** for the trunk (e.g., `Asterisk`).
2. Select **Static Endpoint** as the trunk **Type**.
3. In **Host**, enter the public IP address of the Asterisk server or a
   domain name that resolves to it.
4. Select the signaling **Transport** and enter the corresponding
   **Port** used by Asterisk. The standard port is `5060` for UDP or
   TCP and `5061` for TLS. If Asterisk uses a custom listening port,
   enter that port instead.
5. Set **User Part of R-URI** to `{DID}`. This placeholder inserts the
   called DID in E.164 format into the user part of the Request-URI.

[![Configuring a Static Endpoint inbound SIP trunk in the DIDWW User Panel](../../_images/configure-static-endpoint.webp)](../../_images/configure-static-endpoint.webp)


Static Endpoint trunk settings.


Dynamic Registration

1. Enter a descriptive **Name** for the trunk (e.g., `Asterisk`).
2. Select **Dynamic Registration** as the trunk **Type**.
3. Enable **Use DID in R-URI** so the called DID replaces the registered
   contact user part in inbound requests.

[![Selecting Dynamic Registration and Use DID in R-URI for an inbound SIP trunk](../../_images/configure-dynamic-registration.webp)](../../_images/configure-dynamic-registration.webp)


Dynamic Registration trunk settings.

### Step 3: Click Create and Save Inbound SIP Trunk Configuration

When all required fields in the Create Inbound SIP Trunk form are filled,
click **Create** to save the trunk.

Note

For advanced SIP trunk configuration, see
[Advanced Inbound SIP Trunk documentation](../../voice/inbound-trunks/creating-a-new-sip-trunk.html#user-panel-sipin-trunk-advanced-configuration).

Static Endpoint

[![Saving the Static Endpoint inbound SIP trunk](../../_images/save-static-inbound-trunk.webp)](../../_images/save-static-inbound-trunk.webp)


Creating the Static Endpoint inbound SIP trunk.


Dynamic Registration

[![Saving the Dynamic Registration inbound SIP trunk](../../_images/save-dynamic-inbound-trunk.webp)](../../_images/save-dynamic-inbound-trunk.webp)


Creating the Dynamic Registration inbound SIP trunk.

### Step 4: Copy Inbound Trunk Credentials (Dynamic Registration Only)

For a Dynamic Registration trunk, open the created trunk and copy its
generated username, password, and registration endpoint. You will use these
values in `pjsip.conf`. Static Endpoint trunks do not use registration
credentials.

[![Viewing generated Dynamic Registration credentials and endpoints](../../_images/inbound-registration-credentials.webp)](../../_images/inbound-registration-credentials.webp)


Dynamic Registration credentials and endpoints.

### Step 5: Assign inbound SIP trunk to your DID numbers

After creating the Inbound SIP Trunk for Asterisk, assign it to the DID
number(s) that will deliver incoming calls to Asterisk.

1. In the DIDWW User Panel, go to **Phone Numbers > My Numbers**.
2. Select the DID number(s) you want to assign to the inbound SIP trunk.
3. At the bottom of the page, click **Batch Actions > Update Trunks**.

[![Selecting DID numbers and Update Trunks from the Batch Actions menu](../../_images/select-dids-update-trunks.webp)](../../_images/select-dids-update-trunks.webp)


Selecting **Update Trunks** from the Batch Actions menu.

4. From the dropdown menu, choose the **Asterisk SIP trunk** you created
   earlier.
5. Click **Confirm** to apply the changes.

[![Assigning a SIP trunk to DID numbers](../../_images/assign-inbound-trunk.webp)](../../_images/assign-inbound-trunk.webp)


Assigning the newly created SIP trunk to the selected DID(s).

---

## 2. Configure outbound voice trunk

Configure an Outbound SIP Trunk in the DIDWW User Panel to allow Asterisk to
place outbound calls through DIDWW. This trunk provides the SIP credentials
and routing settings required for outbound calls to external phone numbers.

### Before you begin

Access to **DIDWW Outbound Trunks** is required for making outbound calls. See [Get Access to DIDWW Outbound
Termination](../../voice/outbound-trunks/get-access.html#user-panel-voice-out-get-access).

### Step 1: Create New Outbound SIP Trunk

1. In the [DIDWW User Panel](https://my.didww.com/#/trunks), go to **Voice >
   Outbound Trunks**.
2. Click **Create New**.

[![Creating a new outbound SIP trunk](../../_images/create-outbound-sip-trunk.png)](../../_images/create-outbound-sip-trunk.png)


Creating a new outbound SIP trunk.

### Step 2: Configure Authentication

1. Update the **Friendly Name** (e.g., `Asterisk`).
2. Keep the default **Credentials & IP-based** authentication method selected.
   The SIP digest credentials (username and password) will be provided after
   the trunk is created.
3. In **Allowed SIP IP addresses**, enter the public IP address or subnet from
   which Asterisk will send outbound SIP traffic.
4. Configure **Allowed CLI(s)** and the other termination settings required by
   your deployment.

Note

Make sure you add the correct public IP address or subnet so that outbound
calls are accepted by DIDWW.

[![Configuring credentials and allowed SIP addresses for an outbound trunk](../../_images/configure-outbound-trunk.webp)](../../_images/configure-outbound-trunk.webp)


Entering allowed SIP IP addresses for outbound authentication.

### Step 3: Click Create and Save Outbound SIP Trunk Configuration

When all required fields in the Create Outbound SIP Trunk are filled, click
**Create** to save your outbound SIP trunk.

Note

For advanced outbound SIP trunk configuration, see
[Outbound SIP Trunk Guide](../../voice/outbound-trunks/how-to-guides/index.html#outbound-trunk-create).

[![Saving the outbound SIP trunk](../../_images/save-outbound-trunk.webp)](../../_images/save-outbound-trunk.webp)


Outbound SIP trunk created and ready for use.

### Step 4: View Outbound Trunk Credentials

After the outbound trunk is created, you can view its credentials by selecting
the key icon in the Credentials column on the Outbound Trunks page.

1. Go to **Voice > Outbound Trunks**.
2. Locate your outbound trunk and click the **key icon** in the **Credentials**
   column.

[![Selecting the credentials icon for the Asterisk outbound trunk](../../_images/open-outbound-trunk-credentials.webp)](../../_images/open-outbound-trunk-credentials.webp)


Opening the Asterisk outbound trunk credentials.

3. The trunk credentials will appear, showing the **Username** and **Password**
   (click the **eye icon** to reveal the password).
4. Copy and securely store these credentials. You will need them when
   configuring outbound calling in Asterisk.

[![Viewing the Asterisk outbound trunk username, password, and hostnames](../../_images/outbound-trunk-credentials.webp)](../../_images/outbound-trunk-credentials.webp)


Asterisk outbound trunk credentials and hostnames.

---

## 3. Configure Asterisk

Configure Asterisk to use the DIDWW inbound and outbound trunks created in the
previous sections. Add the required PJSIP transport, trunk, and dialplan
settings for your deployment.

### Before you begin

Asterisk 20 or later is required with `chan_pjsip` and `res_pjsip` loaded. See the official [Configuring
res\_pjsip](https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/) documentation.

Note

The examples use `/etc/asterisk/pjsip.conf` and
`/etc/asterisk/extensions.conf`. Replace all uppercase placeholder values
before applying the configuration.

For an explanation of the transport,
endpoint, authentication, AoR, registration, and identify objects, see the
official [PJSIP Configuration Sections and
Relationships](https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/PJSIP-Configuration-Sections-and-Relationships/)
documentation.

### Step 1: Configure SIP trunks (pjsip.conf)

Complete the core trunk configuration first. Then add only the optional
settings required by the Asterisk deployment.

#### Core trunk configuration

In `/etc/asterisk/pjsip.conf`, configure the SIP transport and the inbound
and outbound DIDWW trunks. These settings establish the core SIP signaling
and media configuration required for calling through DIDWW.

##### Configure SIP transport

Define the signaling transport in `pjsip.conf`. For inbound calling, the
transport and port must match the DIDWW inbound trunk settings. Skip this step
if `pjsip.conf` already contains the required transport.

UDP

Add the UDP transport:

```
[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0:5060
```


TCP

Add the TCP transport:

```
[transport-tcp]
type = transport
protocol = tcp
bind = 0.0.0.0:5060
```


TLS

Add the TLS transport:

```
[transport-tls]
type = transport
protocol = tls
bind = 0.0.0.0:5061
method = tlsv1_2
cert_file = /etc/asterisk/keys/asterisk.crt
priv_key_file = /etc/asterisk/keys/asterisk.key
verify_server = yes
ca_list_file = /etc/ssl/certs/ca-certificates.crt
```

Replace `cert_file` and `priv_key_file` with the paths to the
Asterisk TLS certificate and private key. Update `ca_list_file` if the
system CA bundle uses a different path.

Note

The remaining examples use UDP. If Asterisk is behind NAT, complete [Configure NAT and firewall rules](#asterisk-nat).

For Asterisk transport selection rules and additional examples, see the
official [PJSIP Transport
Selection](https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/PJSIP-Transport-Selection/)
documentation.

##### Configure inbound trunk

Add the configuration that matches the inbound trunk type selected in the
DIDWW User Panel to `pjsip.conf`.

Static Endpoint

Add the inbound endpoint and `identify` object:

```
[didww-in]
type = endpoint
transport = transport-udp
context = from-didww
disallow = all
allow = alaw,ulaw
dtmf_mode = rfc4733
direct_media = no

[didww-in-identify]
type = identify
endpoint = didww-in
match = 46.19.209.14
match = 46.19.210.14
match = 46.19.212.14
match = 46.19.213.14
match = 46.19.214.14
match = 46.19.215.14
match = 185.238.173.14
```

Replace `transport-udp` with `transport-tcp` or `transport-tls`
when using TCP or TLS. **Preferred Server** is set to **Auto** by
default, so add all addresses listed under [SIP Signaling
Addresses](../../voice/inbound-trunks/technical-data/sip.html#service-did-sip).

Note

- The `identify` object maps incoming DIDWW traffic to the
  `didww-in` endpoint. No `aor` or `registration` object is
  required.
- This guide uses IPv4. For IPv6, use the DIDWW addresses under
  [SIP Signaling Addresses](../../voice/inbound-trunks/technical-data/sip.html#service-did-sip) and configure an
  IPv6 [PJSIP
transport](https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/PJSIP-Transport-Selection/).


Dynamic Registration

Add the inbound endpoint and registration using the credentials provided
by DIDWW:

```
[didww-reg-auth]
type = auth
auth_type = userpass
username = INBOUND_TRUNK_USERNAME
password = INBOUND_TRUNK_PASSWORD

[didww-reg]
type = registration
transport = transport-udp
outbound_auth = didww-reg-auth
server_uri = sip:sip.didww.com
client_uri = sip:INBOUND_TRUNK_USERNAME@sip.didww.com
retry_interval = 60
expiration = 3600
line = yes
endpoint = didww-in

[didww-in]
type = endpoint
transport = transport-udp
context = from-didww
disallow = all
allow = alaw,ulaw
dtmf_mode = rfc4733
direct_media = no
```

Replace `transport-udp` with `transport-tcp` when using TCP. When
using TLS, set `transport = transport-tls` and include port `5061`
in both registration URIs:

```
server_uri = sip:sip.didww.com:5061\;transport=tls
client_uri = sip:INBOUND_TRUNK_USERNAME@sip.didww.com:5061\;transport=tls
```

`sip.didww.com` selects the registration location automatically. See
[SIP Registrars](../../voice/inbound-trunks/technical-data/sip.html#service-did-sip) to use a regional hostname.

Note

- Replace `INBOUND_TRUNK_USERNAME` and
  `INBOUND_TRUNK_PASSWORD` with the credentials copied from the
  DIDWW User Panel.
- `line = yes` and `endpoint = didww-in` associate incoming calls
  with the `didww-in` endpoint. No `identify` object or signaling
  IP address list is required.

For both trunk types, the called DID is sent to the `from-didww` dialplan
context in E.164 format.

The inbound examples use G.711 A-law and G.711 µ-law. See
[Supported codecs](../../voice/inbound-trunks/technical-data/sip.html#service-did-sip-codecs) to use other inbound codecs.

##### Configure outbound trunk

Add the outbound trunk to `pjsip.conf`:

```
[didww-out-auth]
type = auth
auth_type = userpass
username = OUTBOUND_TRUNK_USERNAME
password = OUTBOUND_TRUNK_PASSWORD

[didww-out]
type = aor
contact = sip:any.out.didww.com
qualify_frequency = 60

[didww-out]
type = endpoint
transport = transport-udp
disallow = all
allow = alaw,ulaw,g729
dtmf_mode = rfc4733
direct_media = no
outbound_auth = didww-out-auth
aors = didww-out
from_user = AUTHORIZED_CALLER_ID
```

Note

- Replace `OUTBOUND_TRUNK_USERNAME` and `OUTBOUND_TRUNK_PASSWORD` with
  the credentials copied from the DIDWW User Panel.
- Replace `AUTHORIZED_CALLER_ID` with a caller ID allowed by the outbound
  trunk's CLI settings, for example, `12025550123`.
- Send destination and caller ID numbers in E.164 format. See
  [Outbound dialing](../../voice/outbound-trunks/routing-dialing/outbound-dialing.html).

In `/etc/asterisk/pjsip.conf`, add the following setting to the existing
`type = endpoint` object for each internal PJSIP endpoint that is allowed to
make outbound calls. The `outbound` dialplan context is configured in
[Step 2](#asterisk-configure-dialplan).

```
context = outbound
```

The example uses UDP and the DIDWW anycast endpoint
`any.out.didww.com`. Anycast uses network routing to select a DIDWW point of
presence. See [SIP Protocol Details](../../voice/outbound-trunks/outbound-sip-information.html#services-voice-out) to
use a
regional endpoint instead.

For TCP, replace
`transport-udp` with `transport-tcp`. For TLS, use `transport-tls` and a
regional endpoint on port `5061`. Replace the `contact` and `transport`
values with, for example:

```
; In the didww-out aor object
contact = sip:nyc.us.out.didww.com:5061\;transport=tls

; In the didww-out endpoint object
transport = transport-tls
```

The outbound example uses G.711 A-law, G.711 µ-law, and G.729. See
[SIP Protocol Details](../../voice/outbound-trunks/outbound-sip-information.html#services-voice-out) to use other outbound codecs.

Note

In Asterisk PJSIP, `dtmf_mode = rfc4733` configures the RTP
telephone-event method commonly referred to as RFC 2833. See
[DTMF transport methods](../../voice/inbound-trunks/technical-data/sip.html#service-did-sip-dtmf).

#### Optional trunk configuration

Use only the sections required by your deployment. Configure NAT when Asterisk
is behind NAT. Media encryption and T.38 fax are optional.

##### Configure NAT and firewall rules

When Asterisk is behind NAT, add the following settings to each transport
object used with DIDWW:

```
external_media_address = PUBLIC_IP_ADDRESS
external_signaling_address = PUBLIC_IP_ADDRESS
local_net = 192.168.0.0/16
```

Replace `PUBLIC_IP_ADDRESS` with the public IP address of the Asterisk server
and adjust `local_net` to the local network in CIDR format.

Allow DIDWW signaling and media traffic through the firewall using the
addresses and ports listed in [General SIP
Information](../../voice/inbound-trunks/technical-data/sip.html#service-did-sip) and [SIP Protocol
Details](../../voice/outbound-trunks/outbound-sip-information.html#services-voice-out).
Allow the configured Asterisk SIP port and the RTP range defined in
`rtp.conf` through the firewall. If the Asterisk server is behind port-based
NAT, forward these ports to the server.

For details about the Asterisk NAT settings, see [Configuring res\_pjsip to work through
NAT](https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Configuring-res_pjsip-to-work-through-NAT/).

##### Configure media encryption

The Asterisk media-encryption method must match the **SRTP Mode** of the
[inbound trunk](../../voice/inbound-trunks/creating-a-new-sip-trunk.html#user-panel-sipin-trunk-media-dtmf) and the **Media
Encryption** setting of the [outbound
trunk](../../voice/outbound-trunks/outbound-trunk-reference.html#outbound-trunk-create-media).

Add one of the following configurations to each DIDWW-facing endpoint.

SRTP SDES

Use TLS signaling to protect the SRTP keys included in the SDP.

```
media_encryption = sdes
```


SRTP DTLS

```
media_encryption = dtls
dtls_auto_generate_cert = yes
dtls_verify = fingerprint
dtls_setup = actpass
```

DIDWW also supports ZRTP key negotiation, but stock Asterisk PJSIP does not.
SRTP requires the `res_srtp` module. See the [Asterisk PJSIP endpoint
options](https://docs.asterisk.org/Asterisk_21_Documentation/API_Documentation/Module_Configuration/res_pjsip/)
and the DIDWW encryption details for [inbound
calling](../../voice/inbound-trunks/technical-data/sip.html#service-did-sip) and [outbound
calling](../../voice/outbound-trunks/outbound-sip-information.html#services-voice-out).

Note

Encryption applies only to the call leg between DIDWW and Asterisk. It is
not end-to-end encryption.

##### Configure T.38 fax

For an inbound DID that supports T.38, add the following setting to the
`didww-in` endpoint:

```
t38_udptl = yes
```

This setting enables T.38 negotiation and does not affect normal voice calls.
The dialplan must route fax calls to a T.38-capable application or endpoint.
If Asterisk relays T.38 to another PJSIP endpoint, enable `t38_udptl` on that
endpoint as well.

When Asterisk is behind NAT, also add `t38_udptl_nat = yes` and allow the
UDPTL port range configured in `udptl.conf`. For T.38 availability and G.711
fax alternatives, see [FAX Services](../../voice/inbound-trunks/technical-data/fax-services.html).

Important

Run `pjsip reload` from the [Asterisk
CLI](https://docs.asterisk.org/Operation/Asterisk-Command-Line-Interface/CLI-Syntax-and-Help-Commands/)
after changing PJSIP endpoints, authentication objects, or registrations.
Transport changes require a complete Asterisk restart.

### Step 2: Configure the dialplan (extensions.conf)

Add the inbound and outbound routing contexts to `extensions.conf`.

```
[from-didww]
exten => INBOUND_DID,1,NoOp(DIDWW inbound call to ${EXTEN})
 same => n,Dial(PJSIP/INTERNAL_ENDPOINT,30)
 same => n,Hangup()

[outbound]
; Send an E.164 destination through the DIDWW outbound trunk.
exten => _X.,1,Set(CALLERID(num)=AUTHORIZED_CALLER_ID)
 same => n,Dial(PJSIP/${EXTEN}@didww-out)
 same => n,Hangup()
```

Replace `INBOUND_DID` with the DID in E.164 format, for example,
`12025550123`; `INTERNAL_ENDPOINT` with the destination PJSIP endpoint,
for example, `1001`; and `AUTHORIZED_CALLER_ID` with a caller ID allowed
by the outbound trunk, for example, `12025550123`.

Warning

Keep the `from-didww` and `outbound` contexts separate. Do not allow
inbound calls to access the `didww-out` endpoint.

For dialplan concepts and `extensions.conf` syntax, see the official
[Asterisk dialplan](https://docs.asterisk.org/Configuration/Dialplan/)
documentation.

Note

Run `dialplan reload` from the [Asterisk
CLI](https://docs.asterisk.org/Operation/Asterisk-Command-Line-Interface/CLI-Syntax-and-Help-Commands/)
after changing `extensions.conf`.

### Step 3: Verify and troubleshoot the configuration

Use the following [Asterisk
CLI](https://docs.asterisk.org/Operation/Asterisk-Command-Line-Interface/CLI-Syntax-and-Help-Commands/)
commands to confirm that Asterisk loaded the configuration and can communicate
with DIDWW:

| Check | Command |
| --- | --- |
| Loaded endpoints | `pjsip show endpoints` |
| Configured transports | `pjsip show transports` |
| IP-based endpoint matches | `pjsip show identifies` |
| Registration state | `pjsip show registrations` |
| Gateway reachability | `pjsip show aors` and `pjsip show contacts` |
| Loaded dialplan contexts | `dialplan show from-didww` and `dialplan show outbound` |
| Live SIP trace | `pjsip set logger on` |
| Force an inbound registration attempt | `pjsip send register didww-reg` |

#### Common issues

Inbound calls return `401` or `404`, or Asterisk reports `No matching endpoint`.

For a Static Endpoint trunk, confirm that the request source is present
in the `identify` object.

For a Dynamic Registration trunk, confirm that `line = yes` and
`endpoint = didww-in` are set on the registration.

An inbound call reaches Asterisk but does not reach the internal endpoint.

Run `dialplan show from-didww` and confirm that the DID received in
the Request-URI matches `INBOUND_DID` in `extensions.conf`.

Confirm that `INTERNAL_ENDPOINT` identifies an available PJSIP endpoint.

A call has one-way audio.

Check `external_media_address`, `local_net`, and the firewall rules.

Confirm that the applicable DIDWW RTP ranges and the Asterisk RTP port
range are allowed.

A call fails with `488` Not Acceptable Here.

Confirm that Asterisk and the corresponding DIDWW trunk have a common
codec.

If media encryption is enabled, confirm that the configured SRTP method
matches the trunk settings.

DTMF input is not recognized.

Confirm that the endpoint uses `dtmf_mode = rfc4733`.

This is the Asterisk PJSIP setting for the RTP telephone-event method
commonly referred to as RFC 2833.

Outbound calls return `403` or repeat `407` challenges.

Confirm the outbound trunk credentials, the public Asterisk address in
Allowed SIP IP addresses, and the caller ID in the trunk's CLI settings.

Asterisk cannot establish a TLS connection.

Confirm that the TLS transport is loaded, the certificate and CA bundle
paths are correct, and the configured DIDWW hostname uses port `5061`.

For outbound TLS, use a regional DIDWW endpoint.

Run `pjsip set logger on` to inspect the failure.

A Dynamic Registration trunk remains `Rejected`.

Confirm that the username in `client_uri` matches the generated trunk
username.

Run `pjsip set logger on` to inspect the authentication challenge and
response.

A Dynamic Registration trunk remains `Unregistered` or cannot reach the registrar.

Confirm that the selected transport is loaded and that the firewall
permits signaling to the configured DIDWW registrar and port.

Confirm that `server_uri` uses the same transport and port.

For additional diagnostic procedures, see the official [Asterisk PJSIP
Troubleshooting
Guide](https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Asterisk-PJSIP-Troubleshooting-Guide/).

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