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.
Full encryption for secure deployments
Optimized for chan_pjsip configs
Credentials in minutes, not days
Engineers who know Asterisk
Copy-paste configs for chan_pjsip. We also support legacy chan_sip.
; 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=no; 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)We fully support encrypted SIP:
transport=transport-tls
media_encryption=sdes
No hidden fees. Scale as you need.
Predictable costs for steady volume
No monthly fees, just usage
Yes, we support both chan_pjsip (recommended) and legacy chan_sip.
G.711 (ulaw/alaw), G.722 for HD voice, G.729, and Opus. G.711 ulaw recommended.
Yes, we support both registration and IP-based authentication. Contact support to whitelist your IPs.
Yes, T.38 fax passthrough is supported. Enable t38_udptl in your pjsip.conf.
Get your SIP credentials in minutes. Our support team is here to help.