SIP Trunking for
Asterisk PBX
Reliable, secure SIP trunks optimized for Asterisk. Works with chan_pjsip and chan_sip. Crystal-clear audio, instant provisioning, and engineers who actually know Asterisk.
TLS + SRTP
Full encryption for secure deployments
PJSIP Ready
Optimized for chan_pjsip configs
Instant Setup
Credentials in minutes, not days
Expert Support
Engineers who know Asterisk
What is an Asterisk SIP Trunk?
An Asterisk SIP trunk is a virtual phone line that connects your Asterisk PBX to the public telephone network (PSTN) over the internet. Instead of expensive PRI circuits ($300–$800/month for 23 channels) or analog lines, SIP trunks route your calls through your existing broadband connection.
With IPComms, your Asterisk SIP trunks have no per-channel fees — no monthly minimums and no contracts. You only pay for the calls you make ($0.009/min) and the phone numbers you use ($1.50/mo).
Why Switch from PRI to SIP?
$300–$800/mo for 23 channels, long provisioning, inflexible
$0/channel, unlimited concurrent calls, instant provisioning
Outbound $0.010/min, inbound $0.009/min, local DIDs $1.50/mo
Port local & toll-free numbers. Use new DIDs immediately while porting.
chan_pjsip vs chan_sip: Which Should You Use?
Asterisk has two SIP channel drivers. Here's what you need to know — IPComms supports both.
chan_pjsip
The modern, actively maintained SIP stack. Default since Asterisk 13.
- ✓Multiple registrations per endpoint
- ✓Native WebRTC support
- ✓Better TLS/SRTP handling
- ✓Actively maintained & updated
- ✓Supported in Asterisk 13–22+
chan_sip
The original SIP driver. Deprecated since Asterisk 17, removed in Asterisk 21.
- ✓Simple sip.conf configuration
- ✓Familiar to long-time admins
- ✗Deprecated — no new features
- ✗Removed in Asterisk 21+
- ✗Limited TLS/SRTP support
Asterisk Version Compatibility
IPComms works with every supported Asterisk release.
| Version | Status | SIP Driver | IPComms |
|---|---|---|---|
| Asterisk 22 | Current | chan_pjsip only | ✓ Supported |
| Asterisk 21 | Standard | chan_pjsip only | ✓ Supported |
| Asterisk 20 LTS | LTS | chan_pjsip + chan_sip | ✓ Supported |
| Asterisk 18 LTS | LTS | chan_pjsip + chan_sip | ✓ Supported |
| Asterisk 13/16 | EOL | chan_pjsip + chan_sip | ✓ Works* |
Easy Configuration
Copy-paste configs for chan_pjsip. We also support legacy chan_sip.
pjsip.conf (Recommended)
; IPComms SIP Trunk - PJSIP
[ipcomms]
type=registration
transport=transport-udp
outbound_auth=ipcomms
server_uri=sip:sip.ipcomms.net
client_uri=sip:YOUR_USER@sip.ipcomms.net
[ipcomms]
type=auth
auth_type=userpass
username=YOUR_USERNAME
password=YOUR_PASSWORD
[ipcomms]
type=aor
contact=sip:sip.ipcomms.net
[ipcomms]
type=endpoint
context=from-trunk
disallow=all
allow=ulaw,alaw,g722
outbound_auth=ipcomms
aors=ipcomms
direct_media=noextensions.conf
; Outbound calls via IPComms
[outbound-ipcomms]
exten => _1NXXNXXXXXX,1,Dial(PJSIP/${EXTEN}@ipcomms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(PJSIP/1${EXTEN}@ipcomms)
exten => _NXXNXXXXXX,n,Hangup()
; Inbound from IPComms
[from-trunk]
exten => _X.,1,NoOp(Inbound: ${CALLERID(num)})
exten => _X.,n,Goto(internal,s,1)Need TLS/SRTP?
We fully support encrypted SIP:
transport=transport-tls
media_encryption=sdes
View complete TLS/SRTP configuration guide →
Simple Pricing
No hidden fees. Scale as you need.
Metered
Predictable costs for steady volume
Pay-As-You-Go
No monthly fees, just usage
Asterisk SIP Trunking FAQ
Common questions about using SIP trunks with Asterisk PBX.
What is an Asterisk SIP trunk?
An Asterisk SIP trunk is a virtual phone line that connects your Asterisk PBX to the public telephone network (PSTN) over the internet using the Session Initiation Protocol. Instead of physical PRI or analog lines, SIP trunks route calls through your broadband connection, reducing costs by 40–60% while adding flexibility and scalability.
Should I use chan_pjsip or chan_sip?
chan_pjsip is recommended for all new Asterisk installations. It is the actively maintained SIP channel driver since Asterisk 13 and supports modern features like multiple registrations per endpoint, WebRTC, and better TLS/SRTP handling. chan_sip is deprecated since Asterisk 17 and removed in Asterisk 21. IPComms supports both. See our PJSIP configuration guide for details.
How much do Asterisk SIP trunks cost?
IPComms Asterisk SIP trunks are free — there are no per-channel or per-trunk fees. You only pay for usage: outbound calls to the USA/Canada cost $0.010/min, inbound local calls cost $0.009/min, and local DIDs are $1.50/month. No contracts and no monthly minimums. See full pricing.
What codecs do you support?
IPComms supports G.711 (ulaw/alaw), G.722 for HD voice, G.729, and Opus. G.711 ulaw is recommended for the best compatibility and call quality.
Can I use IP authentication instead of registration?
Yes, IPComms supports both registration-based and IP-based authentication for Asterisk SIP trunks. Contact support to whitelist your static IP addresses.
What versions of Asterisk are supported?
IPComms works with all currently supported Asterisk versions including Asterisk 18 LTS, 20 LTS, 21, and 22. We also support older versions like Asterisk 13 and 16, though we recommend upgrading. See our Asterisk installation guide to get started.
Do you support T.38 fax?
Yes, T.38 fax passthrough is fully supported. Enable t38_udptl=yes in your pjsip.conf endpoint configuration to use it.
Can I port my existing phone numbers to IPComms?
Yes. IPComms supports number porting for local and toll-free numbers across the United States. The process typically takes 7–14 business days for local numbers. You can start using IPComms immediately with new numbers while your existing numbers are being ported.
Asterisk Guides
Step-by-step tutorials from our Asterisk experts.
Asterisk PJSIP Configuration
Complete guide to pjsip.conf settings, reload commands, and troubleshooting.
Asterisk PJSIP Trunk Setup
Connect Asterisk to IPComms SIP trunks step-by-step with PJSIP.
Install Asterisk on Debian 13
Build Asterisk from source on Debian Trixie with free SIP trunks.
Asterisk Call Queues
Configure queues.conf with strategies, announcements, and timeouts.
Fix One-Way Audio
Solve NAT, firewall, and RTP issues causing one-way or no audio.
SIP Trunk Failover
Design redundant voice paths for business continuity.
Best Free Softphones
Top 10 free softphone apps for testing your Asterisk SIP trunk.
Ready to connect your Asterisk PBX?
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