IP PBX (web UI on Asterisk)
Configure the SIP Trunk on Issabel
Set up the PortalPBX SIP trunk in Issabel from the web UI: trunk, inbound route per DID and outbound route.
Before you start
- Administrator access to the Issabel web UI.
- Trunk details provided by PortalPBX (server, username, password, DID).
- Internet access to the SIP server and the RTP range open in the firewall.
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
Create the SIP trunk
Go to PBX → PBX Configuration → Trunks → Add SIP Trunk. Give it a name (e.g. portalpbx) and set the outbound Caller ID to your DID.
- 2
PEER details (outgoing)
In “PEER Details” enter the trunk parameters with your data:
type=peer host=<servidor-sip> username=<usuario> secret=<contraseña> fromuser=<usuario> fromdomain=<servidor-sip> insecure=port,invite qualify=yes nat=force_rport,comedia disallow=all allow=alaw,ulaw context=from-trunk - 3
Registration string
If your trunk uses registration, fill in “Register String”:
<usuario>:<contraseña>@<servidor-sip>/<DID> - 4
Inbound route
PBX → Inbound Routes → add a route with the DID and choose the destination (IVR, queue, ring group or extension).
- 5
Outbound route
PBX → Outbound Routes → create dial patterns (landline, mobile, international) and select the portalpbx trunk.
- 6
Apply and verify
Apply changes and check the status in the Asterisk console:
asterisk -rx "sip show registry" asterisk -rx "sip show peers"
Recommendations
- In Issabel SIP Settings, set the external IP and local network if you are behind NAT.
- 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
401 / 403
Wrong username/password or unauthorized IP. Check the details and the authentication mode.
408 / Request Timeout
The server does not answer: firewall, DNS or blocked port.
488 Not Acceptable
No codec in common. Enable alaw/ulaw.
Audio en una vía
NAT or SIP ALG. Check external IP, local network and disable SIP ALG.
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.