Asterisk SIP Trunking

Configure your Asterisk PBX to connect with IPComms SIP trunking services using PJSIP or chan_sip.

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Connection Details PJSIP Configuration chan_sip Configuration Dialplan Examples TLS/Secure Configuration Troubleshooting

1 Connection Details

Use the following settings to configure your Asterisk trunk. Find your credentials in the IPComms Portal under SIP Trunking.

Setting Standard SIP TLS/SRTP
SIP Server [subdomain].s1.ipcomms.net [subdomain].s1.ipcomms.net
Port 5060 (UDP) 5061 (TCP/TLS)
Username [Your Trunk Username]
Password [Your Trunk Password]
Auth Username [Same as Username]
Outbound Proxy [subdomain].s1.ipcomms.net
Codecs G.711 (ulaw/alaw), G.729
DTMF Mode RFC 2833

Note: Replace [subdomain] with your assigned subdomain from the portal (e.g., abc1234.s1.ipcomms.net).

2 PJSIP Configuration

PJSIP is the recommended SIP channel driver for Asterisk 13+. Add the following to your pjsip.conf file:

Standard SIP (UDP Port 5060)

; ============================================
; IPComms SIP Trunk - PJSIP Configuration
; File: /etc/asterisk/pjsip.conf
; ============================================

; Transport for standard SIP
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060

; IPComms Trunk Registration
[ipcomms]
type=registration
transport=transport-udp
outbound_auth=ipcomms-auth
server_uri=sip:YOUR_SUBDOMAIN.s1.ipcomms.net
client_uri=sip:YOUR_USERNAME@YOUR_SUBDOMAIN.s1.ipcomms.net
contact_user=YOUR_USERNAME
retry_interval=60
forbidden_retry_interval=600
expiration=3600

; Authentication
[ipcomms-auth]
type=auth
auth_type=userpass
username=YOUR_USERNAME
password=YOUR_PASSWORD

; AOR (Address of Record)
[ipcomms-aor]
type=aor
contact=sip:YOUR_SUBDOMAIN.s1.ipcomms.net:5060
qualify_frequency=60
qualify_timeout=5.0

; Endpoint Configuration
[ipcomms-endpoint]
type=endpoint
transport=transport-udp
context=from-ipcomms
disallow=all
allow=ulaw
allow=alaw
allow=g729
outbound_auth=ipcomms-auth
aors=ipcomms-aor
direct_media=no
dtmf_mode=rfc4733
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
from_domain=YOUR_SUBDOMAIN.s1.ipcomms.net
from_user=YOUR_USERNAME

; Identify incoming calls from IPComms
[ipcomms-identify]
type=identify
endpoint=ipcomms-endpoint
match=YOUR_SUBDOMAIN.s1.ipcomms.net

Replace the following values:
YOUR_SUBDOMAIN - Your IPComms subdomain (e.g., abc1234)
YOUR_USERNAME - Your trunk username from the portal
YOUR_PASSWORD - Your trunk password from the portal

3 chan_sip Configuration (Legacy)

Note: chan_sip is deprecated in newer versions of Asterisk. We recommend using PJSIP for new installations. This configuration is provided for legacy systems.

Add the following to your sip.conf file:

Standard SIP (UDP Port 5060)

; ============================================
; IPComms SIP Trunk - chan_sip Configuration
; File: /etc/asterisk/sip.conf
; ============================================

; General Settings (add to [general] section)
[general]
register => YOUR_USERNAME:YOUR_PASSWORD@YOUR_SUBDOMAIN.s1.ipcomms.net/YOUR_USERNAME

; IPComms Trunk Definition
[ipcomms]
type=peer
host=YOUR_SUBDOMAIN.s1.ipcomms.net
port=5060
username=YOUR_USERNAME
secret=YOUR_PASSWORD
fromuser=YOUR_USERNAME
fromdomain=YOUR_SUBDOMAIN.s1.ipcomms.net
context=from-ipcomms
insecure=port,invite
qualify=yes
qualifyfreq=60
disallow=all
allow=ulaw
allow=alaw
allow=g729
dtmfmode=rfc2833
nat=force_rport,comedia
canreinvite=no
directmedia=no
trustrpid=yes
sendrpid=yes

4 TLS/SRTP Secure Configuration

For encrypted calls, use TLS for signaling and SRTP for media encryption.

PJSIP with TLS (Port 5061)

; ============================================
; IPComms SIP Trunk - PJSIP TLS Configuration
; File: /etc/asterisk/pjsip.conf
; ============================================

; TLS Transport
[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
cert_file=/etc/asterisk/keys/asterisk.pem
ca_list_file=/etc/asterisk/keys/ca.crt
method=tlsv1_2
verify_server=no

; IPComms TLS Trunk Registration
[ipcomms-tls]
type=registration
transport=transport-tls
outbound_auth=ipcomms-auth
server_uri=sips:YOUR_SUBDOMAIN.s1.ipcomms.net:5061
client_uri=sips:YOUR_USERNAME@YOUR_SUBDOMAIN.s1.ipcomms.net:5061
contact_user=YOUR_USERNAME
retry_interval=60
expiration=3600

; Authentication (same as standard)
[ipcomms-auth]
type=auth
auth_type=userpass
username=YOUR_USERNAME
password=YOUR_PASSWORD

; TLS AOR
[ipcomms-aor-tls]
type=aor
contact=sips:YOUR_SUBDOMAIN.s1.ipcomms.net:5061
qualify_frequency=60
qualify_timeout=5.0

; TLS Endpoint with SRTP
[ipcomms-endpoint-tls]
type=endpoint
transport=transport-tls
context=from-ipcomms
disallow=all
allow=ulaw
allow=alaw
allow=g729
outbound_auth=ipcomms-auth
aors=ipcomms-aor-tls
direct_media=no
dtmf_mode=rfc4733
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
from_domain=YOUR_SUBDOMAIN.s1.ipcomms.net
from_user=YOUR_USERNAME
media_encryption=sdes
media_encryption_optimistic=yes

; Identify for TLS
[ipcomms-identify-tls]
type=identify
endpoint=ipcomms-endpoint-tls
match=YOUR_SUBDOMAIN.s1.ipcomms.net

chan_sip with TLS (Port 5061)

; ============================================
; IPComms SIP Trunk - chan_sip TLS Configuration
; File: /etc/asterisk/sip.conf
; ============================================

; General Settings (add to [general] section)
[general]
tlsenable=yes
tlsbindaddr=0.0.0.0:5061
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1_2

register => tls://YOUR_USERNAME:YOUR_PASSWORD@YOUR_SUBDOMAIN.s1.ipcomms.net:5061/YOUR_USERNAME

; IPComms TLS Trunk Definition
[ipcomms-tls]
type=peer
host=YOUR_SUBDOMAIN.s1.ipcomms.net
port=5061
transport=tls
username=YOUR_USERNAME
secret=YOUR_PASSWORD
fromuser=YOUR_USERNAME
fromdomain=YOUR_SUBDOMAIN.s1.ipcomms.net
context=from-ipcomms
insecure=port,invite
qualify=yes
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
nat=force_rport,comedia
directmedia=no
encryption=yes

5 Dialplan Examples (E.164 Format)

IPComms uses E.164 format for phone numbers. Outbound calls should be sent as +1NPANXXXXXX (e.g., +14045551234).

Outbound Dialplan (extensions.conf)

; ============================================
; IPComms Dialplan - Outbound Calls
; File: /etc/asterisk/extensions.conf
; ============================================

[outbound-ipcomms]
; North American Dialing (NANP) - 10/11 digit calls
; Converts to E.164 format (+1NPANXXXXXX)

; 11-digit dialing (1NPANXXXXXX -> +1NPANXXXXXX)
exten => _1NXXNXXXXXX,1,NoOp(Outbound call to ${EXTEN})
same => n,Set(CALLERID(num)=+1YOUR_DID_NUMBER)
same => n,Dial(PJSIP/+${EXTEN}@ipcomms-endpoint,60)
same => n,Hangup()

; 10-digit dialing (NPANXXXXXX -> +1NPANXXXXXX)
exten => _NXXNXXXXXX,1,NoOp(Outbound call to ${EXTEN})
same => n,Set(CALLERID(num)=+1YOUR_DID_NUMBER)
same => n,Dial(PJSIP/+1${EXTEN}@ipcomms-endpoint,60)
same => n,Hangup()

; 7-digit local dialing (NXXXXXX -> +1AREACODENXXXXXX)
; Replace AREACODE with your local area code
exten => _NXXXXXX,1,NoOp(Local call to ${EXTEN})
same => n,Set(CALLERID(num)=+1YOUR_DID_NUMBER)
same => n,Dial(PJSIP/+1AREACODE${EXTEN}@ipcomms-endpoint,60)
same => n,Hangup()

; Toll-Free Numbers (800, 888, 877, 866, 855, 844, 833)
exten => _1800NXXXXXX,1,Dial(PJSIP/+${EXTEN}@ipcomms-endpoint,60)
exten => _1888NXXXXXX,1,Dial(PJSIP/+${EXTEN}@ipcomms-endpoint,60)
exten => _1877NXXXXXX,1,Dial(PJSIP/+${EXTEN}@ipcomms-endpoint,60)
exten => _1866NXXXXXX,1,Dial(PJSIP/+${EXTEN}@ipcomms-endpoint,60)
exten => _1855NXXXXXX,1,Dial(PJSIP/+${EXTEN}@ipcomms-endpoint,60)
exten => _1844NXXXXXX,1,Dial(PJSIP/+${EXTEN}@ipcomms-endpoint,60)
exten => _1833NXXXXXX,1,Dial(PJSIP/+${EXTEN}@ipcomms-endpoint,60)

; Emergency Services (911)
; IMPORTANT: Ensure E911 is properly configured!
exten => 911,1,NoOp(Emergency Call)
same => n,Dial(PJSIP/911@ipcomms-endpoint,60)
same => n,Hangup()

; International Dialing (011 + Country Code + Number)
exten => _011.,1,NoOp(International call to ${EXTEN})
same => n,Set(INTL_NUM=${EXTEN:3})
same => n,Dial(PJSIP/+${INTL_NUM}@ipcomms-endpoint,120)
same => n,Hangup()

Inbound Dialplan

; ============================================
; IPComms Dialplan - Inbound Calls
; File: /etc/asterisk/extensions.conf
; ============================================

[from-ipcomms]
; Context for incoming calls from IPComms trunk

; Handle incoming calls to your DID
; Replace +1YOURDIDNUMBER with your actual DID in E.164 format
exten => +14045551234,1,NoOp(Incoming call from ${CALLERID(num)})
same => n,Set(CDR(accountcode)=ipcomms-inbound)
same => n,Goto(ivr-main,s,1)

; Alternative: Route all incoming to a default extension
exten => _+1NXXNXXXXXX,1,NoOp(Incoming call to ${EXTEN})
same => n,Answer()
same => n,Dial(PJSIP/100,30)
same => n,Voicemail(100@default,u)
same => n,Hangup()

; Catch-all for any unmatched inbound
exten => _.,1,NoOp(Unmatched inbound: ${EXTEN})
same => n,Answer()
same => n,Playback(invalid)
same => n,Hangup()

chan_sip Dialplan Alternative

; For chan_sip users, replace PJSIP with SIP:

; Outbound
exten => _1NXXNXXXXXX,1,Dial(SIP/+${EXTEN}@ipcomms,60)
exten => _NXXNXXXXXX,1,Dial(SIP/+1${EXTEN}@ipcomms,60)

; For TLS trunk
exten => _1NXXNXXXXXX,1,Dial(SIP/+${EXTEN}@ipcomms-tls,60)

E.164 Format Reference

User Dials Sent to IPComms Description
4045551234 +14045551234 10-digit NANP
14045551234 +14045551234 11-digit with leading 1
5551234 +14045551234 7-digit local (prepend +1+areacode)
18005551234 +18005551234 Toll-Free
01144207946xxxx +44207946xxxx International (UK)

6 Troubleshooting

Registration Failed

  • Verify username and password are correct
  • Check that the subdomain matches your portal assignment
  • Ensure UDP port 5060 (or TCP 5061 for TLS) is not blocked
  • Check asterisk -rx "pjsip show registrations"

No Audio (One-Way or No Audio)

  • Ensure direct_media=no is set
  • Enable rtp_symmetric=yes and force_rport=yes
  • Check RTP port range is open (10000-20000 UDP typically)
  • Verify NAT settings if behind a firewall

TLS Connection Issues

  • Ensure certificate files exist and are readable
  • Check tlsv1_2 is supported
  • Verify port 5061 TCP is open
  • Test with openssl s_client -connect YOUR_SUBDOMAIN.s1.ipcomms.net:5061

Calls Not Going Through

  • Check dialplan syntax with asterisk -rx "dialplan show outbound-ipcomms"
  • Verify E.164 format (+1NPANXXXXXX) is being sent
  • Check /var/log/asterisk/full for SIP errors
  • Ensure your account has sufficient balance

Useful CLI Commands

# Check PJSIP registration status
asterisk -rx "pjsip show registrations"

# Check endpoint status
asterisk -rx "pjsip show endpoint ipcomms-endpoint"

# Enable SIP debugging
asterisk -rx "pjsip set logger on"

# Check active channels
asterisk -rx "core show channels"

# Reload PJSIP configuration
asterisk -rx "pjsip reload"

# Reload dialplan
asterisk -rx "dialplan reload"

Related Documentation

Getting Started

Initial setup and account overview

Softphone Setup

Configure mobile and desktop apps

Desk Phone Setup

Yealink, Polycom, Grandstream, Cisco

Need Help?

Our technical support team can assist with trunk configuration and troubleshooting.

Email Support (678) 460-1475