Asterisk

This guide provides technical configurations for integrating Asterisk with didlogic using two different methods: Digest Authentication (Username/Password) and IP Authentication.

Select the section corresponding to your preferred module (chan_sip or chan_pjsip).

Digest Authentication (User/Pass)

Use this method if your Asterisk server does not have a static IP or if you prefer standard registration.

chan_sip Configuration

Edit /etc/asterisk/sip.conf:

[general] 
register => 94986:YOUR_SIP_ACCOUNT_PASSWORD@sip.**.didlogic.net:5060/61354880986
; Register with didlogic SIP trunk

[didlogic_SIP_trunk] ; Name of the SIP trunk 
type=friend ; Choose friend as it is used for both inbound and outbound calls 
host=sip.**.didlogic.net ; Regional SIP gateway (choose one closest to your location) 
defaultuser=94986 ; SIP trunk ID 
fromuser=61354880986 ; Outbound Caller ID 
remotesecret=YOUR_SIP_ACCOUNT_PASSWORD; SIP trunk password 
context=from-siptrunk 
insecure=port,invite 
canreinvite=no 
disallow=all 
allow=ulaw

[11111] 
type=friend 
username=11111 
secret=YOUR_SIP_ACCOUNT_PASSWORD 
host=dynamic 
context=from-internal 
disallow=all 
allow=ulaw 
nat=force_rport,comedia 
canreinvite=no 
qualify=yes

[99999] 
type=friend 
username=99999 
secret=YOUR_SIP_ACCOUNT_PASSWORD 
host=dynamic 
context=from-internal 
disallow=all 
allow=ulaw 
nat=force_rport,comedia 
canreinvite=no 
qualify=yes

info

Find the full list of didlogic proxy servers here: Outbound Calling

To verify registration:

1. Restart your Asterisk server using the following command: sudo systemctl restart asterisk
2. Enter the following command to know if the SIP trunk has been registered properly: asterisk -x "sip show registry"

chan_pjsip Configuration

Edit /etc/asterisk/pjsip.conf:

[transport-udp-main] 
type=transport 
protocol=udp 
bind=0.0.0.0:5060

[auth_didlogic] 
type=auth 
auth_type=userpass 
username=94986 ; SIP trunk ID 
password=YOUR_SIP_ACCOUNT_PASSWORD

[DID_Logic_aor] 
type=aor 
contact=sip:94986@sip.**.didlogic.net:5060

[DID_Logic] 
type=endpoint 
context=from-siptrunk 
disallow=all 
allow=ulaw 
direct_media=no 
from_user=61354880986 ; Outbound Caller ID 
from_domain=sip.**.didlogic.net ; regional SIP gateway 
aors=DID_Logic_aor 
outbound_auth=auth_didlogic

[DID_Logic_identify] 
type=identify 
endpoint=DID_Logic ; Match the didlogic regional SIP gateways using domain names 
match=sip.nl.didlogic.net 
match=sip.uk.didlogic.net 
match=sip.se.didlogic.net 
match=sip.de.didlogic.net 
match=sip.za.didlogic.net 
match=sip.ca.didlogic.net 
match=sip.lax.didlogic.net 
match=sip.nyc.didlogic.net 
match=sip.hk.didlogic.net 
match=sip.sg.didlogic.net 
match=sip.au.didlogic.net 
match=sip.br.didlogic.net

[reg_didlogic] 
type=registration 
transport=transport-udp-main 
outbound_auth=auth_didlogic 
server_uri=sip:sip.**.didlogic.net:5060 
client_uri=sip:94986@sip.**.didlogic.net 
expiration=120 
retry_interval=20 
max_retries=10

[11111] 
type=endpoint 
transport=transport-udp-main 
context=from-internal 
disallow=all 
allow=ulaw 
aors=11111 
auth=11111

[11111] 
type=auth 
auth_type=userpass 
username=11111 
password=YOUR_SIP_ACCOUNT_PASSWORD

[11111] 
type=aor 
max_contacts=2 
contact=sip:[email protected]:5060 ; 192.168.0.105 is internal IP address

To verify registration:

1. Restart your Asterisk server using the following command: sudo systemctl restart asterisk
2. Enter the following command to know if the SIP trunk has been registered properly: asterisk -x "pjsip show registrations"

IP Authentication

Use this method for high-volume traffic. Note that when IP authentication is enabled for the SIP account:

• Registration is unnecessary.
• Authentication relies on trusted IP addresses.
• Replace the host with the didlogic SIP trunk IP.
• Clean up any password-related fields in the configuration.

info

Email [email protected] with your Asterisk server's static IP to enable this on your account.

chan_sip Configuration

Edit /etc/asterisk/sip.conf:

[general] 
; No need for registration as we are using IP-based authentication 
; Remove the 'register' line entirely.
disallow=all                      ; Disallow all codecs by default 
allow=ulaw                        ; Allow ulaw codec (common for VoIP calls) 
nat=force_rport,comedia           ; NAT traversal settings 
canreinvite=no                    ; Disable direct media reinvites
[DID_Logic] 
type=friend                       ; Allows inbound and outbound calls 
host=sip.**.didlogic.net          ; Replace with the DIDLogic-provided IP 
context=from-siptrunk             ; Context for inbound calls 
insecure=port,invite              ; Trust port and INVITE for authentication 
canreinvite=no                    ; Prevent direct media reinvite 
disallow=all                      ; Disallow all codecs 
allow=ulaw                        ; Allow ulaw codec 
qualify=yes                       ; Monitor trunk availability 
directmedia=no                    ; Avoid direct media

chan_pjsip Configuration

Edit /etc/asterisk/pjsip.conf:

[transport-udp-main] 
type=transport 
protocol=udp 
bind=0.0.0.0:5060
; didlogic SIP Trunk using IP Authentication 

[DID_Logic] 
type=endpoint 
context=from-siptrunk 
disallow=all 
allow=ulaw 
direct_media=no 
from_domain=sip.nl.didlogic.net 
t38_udptl=no 
aors=DID_Logic_aor     ; Reference to the AOR below 
auth=none              ; No authentication required (IP-based)

[DID_Logic_aor] 
type=aor 
contact=sip:sip.nl.didlogic.net:5060

[DID_Logic_identify] 
type=identify 
endpoint=DID_Logic ; Match the didlogic regional SIP gateways using domain names match=sip.nl.didlogic.net 
match=sip.uk.didlogic.net 
match=sip.se.didlogic.net 
match=sip.de.didlogic.net 
match=sip.za.didlogic.net 
match=sip.ca.didlogic.net 
match=sip.lax.didlogic.net 
match=sip.nyc.didlogic.net 
match=sip.hk.didlogic.net 
match=sip.sg.didlogic.net 
match=sip.au.didlogic.net 
match=sip.br.didlogic.net

; Internal Extension 11111 
[11111] 
type=endpoint 
transport=transport-udp-main 
context=from-internal 
disallow=all 
allow=ulaw 
aors=11111 
auth=11111 
direct_media=no

[11111] 
type=auth 
auth_type=userpass 
username=11111 
password=87786118384AAss

[11111] 
type=aor max_contacts=2 
contact=sip:[email protected]:5060

; Internal Extension 99999 
[99999] 
type=endpoint 
transport=transport-udp-main 
context=from-internal 
disallow=all 
allow=ulaw 
aors=99999 
auth=99999 
direct_media=no

[99999] 
type=auth 
auth_type=userpass 
username=99999 
password=87786118384AAss

[99999] 
type=aor 
max_contacts=2 
contact=sip:[email protected]:5060

Dialplan & Routing

The dialplan is configured in /etc/asterisk/extensions.conf for both modules. Ensure all dialed numbers follow the E.164 format (e.g., 18004516258).

Outbound Calls

; Example for chan_sip
exten => _X.,1,NoOp(Outbound call through DID_Logic SIP trunk) 
 same => n,Set(CALLERID(num)=61354880986)  ; Set your outbound Caller ID here 
 same => n,Dial(SIP/${EXTEN}@DID_Logic,60) 
 same => n,Hangup()

; Example for chan_pjsip
exten => _X.,1,Set(CALLERID(num)=61354880986) ; Set your outbound Caller ID here 
 same => n,Dial(PJSIP/${EXTEN}@DID_Logic,60,g) 
 same => n,Hangup()

Inbound Calls (didlogic side)

1. Navigate to your numbers on the didlogic portal.
2. Click Edit on the number you wish to set up.
3. Choose SIP URI in the Forward to dropdown.
4. Enter the destination in the following format: +[YourPhoneNumber]@FQDN:port;transport=connection
   • Example: [email protected]:5060;transport=udp
5. Click Add, and it should appear below. Ensure this destination is at the top of the list if you already have other destinations configured.

Inbound Calls (Asterisk side)

In the chan_sip module:

[general] 
static=yes 
writeprotect=no 
clearglobalvars=no

[globals] 
; Define any global variables if needed

[from-internal] 
; Internal extensions for dialing within the system 
exten => 1001,1,Dial(SIP/11111,20) ; Route call to SIP/11111, ring for 20 seconds 
exten => 1002,1,Dial(SIP/99999,20) ; Route call to SIP/99999, ring for 20 seconds

[from-siptrunk] 
; ============================= 
; Inbound Call Handling for DID Logic 
; =============================

; Handle inbound calls for DID 61354880986 without the '+' prefix 
exten => 61354880986,1,NoOp(Inbound call for DID 61354880986) 
 same => n,Dial(SIP/11111,60)         ; Route call to extension 11111, ring for 60 seconds 
 same => n,Voicemail(1001@default,u)  ; Send to voicemail 1001 (unavailable message) if unanswered 
 same => n,Hangup()
 
; Handle inbound calls for DID 61354880986 with the '+' prefix 
exten => +61354880986,1,NoOp(Inbound call for DID +61354880986) 
 same => n,Dial(SIP/11111,60)         ; Route call to extension 11111, ring for 60 seconds 
 same => n,Voicemail(1001@default,u)  ; Send to voicemail 1001 (unavailable message) if unanswered 
 same => n,Hangup()
 
; ============================= 
; Catch-All for Unknown or Invalid Calls 
; ============================= 
exten => _.,1,NoOp(Unknown inbound call received) 
 same => n,Hangup()

In the chan_pjsip module:

[general] 
static=yes 
writeprotect=no 
clearglobalvars=no

[globals] 
; Define any global variables here if needed

[from-internal] 
; ============================= 
; Internal Extension Dialing 
; ============================= 
exten => 1001,1,Dial(PJSIP/11111,20) ; Route to endpoint 11111, ring for 20 seconds 
exten => 1002,1,Dial(PJSIP/99999,20) ; Route to endpoint 99999, ring for 20 seconds

[from-siptrunk] 
; ============================= 
; Inbound Call Handling from DID Logic 
; =============================

; Handle inbound calls for DID 61354880986 without the '+' prefix 
exten => 61354880986,1,Dial(PJSIP/11111,60)  ; Route to endpoint 11111, ring for 60 seconds 
 same => n,Voicemail(1001@default,u)         ; Send to voicemail 1001 (unavailable message) 
 same => n,Hangup()
 
; Handle inbound calls for DID 61354880986 with the '+' prefix 
exten => +61354880986,1,Dial(PJSIP/11111,60) ; Route to endpoint 11111, ring for 60 seconds 
 same => n,Voicemail(1001@default,u)         ; Send to voicemail 1001 (unavailable message) 
 same => n,Hangup()
 
; ============================= 
; Catch-All for Unhandled Inbound Calls 
; ============================= 
exten => _X.,1,NoOp(Unhandled inbound call to ${EXTEN}) ; Log unhandled calls 
 same => n,Hangup()

Firewall & Audio (RTP)

To prevent "one-way audio" issues, you must allow RTP traffic.

1. Configure Asterisk: Edit /etc/asterisk/rtp.conf:

   [general]
   rtpstart=10000
   rtpend=20000

   Save the file and restart Asterisk with the following command: sudo systemctl restart asterisk

2. Update OS Firewall (Ubuntu/UFW):

   sudo ufw allow 10000:20000/udp
   sudo ufw reload

Troubleshooting Summary

Issue	Cause	Solution
Registration Fail	Wrong Gateway	Double-check your regional gateway (e.g., sip.nyc.didlogic.net).
No Audio	NAT/Firewall	Ensure UDP ports 10000-20000 are open.
Max Call Rate Exceeded	Call rate for the call destination exceeds the max call rate set for your account.	Contact [email protected] to increase the max call rate
402 Payment Req	Low Balance	Refill account balance in the portal.
Wrong Dialing Format	The dialed number may be in an incorrect format.	Ensure the dialing format follows the E.164 standard. For example, a call to a US number should be dialed as 18004516258, not 8004516258 or 08004516258.