Asterisk with PJSIP (config files)
Configure the SIP Trunk on Asterisk
Manual PortalPBX trunk setup in Asterisk 16+ with chan_pjsip: registration, authentication, endpoint and dialplan.
Before you start
- Asterisk 16 or later with chan_pjsip and root SSH access.
- Trunk details provided by PortalPBX.
- A backup of /etc/asterisk before editing.
Your trunk details
PortalPBX sends them when the trunk is activated. Replace each placeholder with your real value.
- SIP server (host or domain)
- <servidor-sip>
- Port and transport
- <puerto> · UDP / TCP / TLS
- SIP username
- <usuario>
- SIP password
- <contraseña>
- Assigned DID number(s)
- <DID>
- Authentication mode
- Registro / IP
- Enabled codecs
- G.711 (alaw/ulaw) · G.729*
* G.729 only if enabled on your trunk.
Step by step
- 1
Transport
In pjsip.conf define the transport; behind NAT add the public IP and local network:
[transport-udp] type=transport protocol=udp bind=0.0.0.0 ;external_media_address=<ip-publica> ;external_signaling_address=<ip-publica> ;local_net=192.168.0.0/16 - 2
Registration and authentication
Add the registration (if your trunk uses it) and the credentials:
[portalpbx-reg] type=registration outbound_auth=portalpbx-auth server_uri=sip:<servidor-sip> client_uri=sip:<usuario>@<servidor-sip> retry_interval=60 [portalpbx-auth] type=auth auth_type=userpass username=<usuario> password=<contraseña> - 3
AOR, endpoint and identify
Define the trunk destination and endpoint:
[portalpbx] type=aor contact=sip:<servidor-sip> qualify_frequency=60 [portalpbx] type=endpoint transport=transport-udp context=from-portalpbx disallow=all allow=alaw,ulaw outbound_auth=portalpbx-auth aors=portalpbx from_user=<usuario> from_domain=<servidor-sip> rtp_symmetric=yes force_rport=yes rewrite_contact=yes [portalpbx] type=identify endpoint=portalpbx match=<servidor-sip> - 4
Dialplan
In extensions.conf: inbound per DID and outbound through the trunk:
[from-portalpbx] exten => <DID>,1,Dial(PJSIP/100,30) same => n,Hangup() [salida] exten => _X.,1,Set(CALLERID(num)=<DID>) same => n,Dial(PJSIP/${EXTEN}@portalpbx,60) same => n,Hangup() - 5
Reload and verify
Reload and check registration and reachability:
asterisk -rx "pjsip reload" asterisk -rx "pjsip show registrations" asterisk -rx "pjsip show endpoint portalpbx"
Recommendations
- To debug, temporarily enable SIP logging: pjsip set logger on.
- Prefer G.711 alaw/ulaw; use G.729 only if it is enabled on your trunk and supported by your equipment.
- Disable SIP ALG on your router or firewall: it is the most common cause of one-way audio and dropped registrations.
- If your PBX is behind NAT, set the public IP and local network, and open the RTP range your system uses.
- Restrict SIP access with a firewall and use long extension passwords to prevent toll fraud.
- Dial numbers in the format stated in your activation sheet (national or international) and check the caller ID.
Common errors
Rejected / 401
Wrong credentials or client_uri.
No matching endpoint
The identify section does not match the source of inbound calls.
488 Not Acceptable
No codec match: check allow=.
Audio en una vía
Check external_media_address, local_net and rtp_symmetric.
Need a hand?
We configure it with you, remotely
If the manual setup does not work, contact support or open a ticket. We review registration, routes, codecs and NAT with you through a secure remote session.