Follow

ASTERISK PBX

This guide provides the settings to manually configure the SIP registration for an Asterisk device with SIPTRUNK. We recommend you create multiple redundant configurations to register the SIP trunk to each of our servers at gw1.siptrunk.com, gw2.siptrunk.com, gw4.siptrunk.com, and gw5.siptrunk.com. 

 

1) Create the SIP trunk name "xxxxxxxxxxGW#" for each registration, where xxxxxxxxxx is your SIPTRUNK SIP trunk number and # is 1 for gw1, 2 for gw2, 4 for gw4, and 5 for gw5. 

 

2. Add the "Peer Details":

  • In the "host" line, add gw#.siptrunk.com

  • In the "fromdomain" line, enter gw#.siptrunk.com

  • Insert the SIP trunk number into the "username" line.

  • Insert the SIP trunk password in the "secret" line.

 

3) Your settings should read as follows:

  • type=peer

  • insecure=port,invite

  • host=gw#.siptrunk.com

  • port=5060

  • dtmfmode=rfc2833

  • canreinvite=no

  • disallow=all

  • allow=ulaw

  • qualify=yes

  • qualifyfreq=30

  • nat=yes

  • trustrpid=yes

  • fromdomain=gw#.siptrunk.com

  • username="______________"

  • secret="______________"

  • context=from-trunk

  • rfc2833compensate=yes

  • session-timers=refuse

 

4) Add the Register String (xxxxxxxxxx is your SIPTRUNK trunk number, yyyyyyyyyyyy is the SIP trunk password, and # is for the server gateway (gw)). 

  • xxxxxxxxxx:yyyyyyyyyyyy@gw#.siptrunk.com

 

Outbound Routing

5) Create a Route name SIPTRUNK_xxxxxxxxxx where xxxxxxxxxx is your SIPTRUNK trunk.

 

NOTE: SIPTRUNK requires 1+10-digit dialing within NANPA, N11/N33 for Emergency and Directory Services, and no prefix for international dialing (only country code + number). For more information, refer to this article on Setting Up Outbound Calling

 

6) SIPTRUNK recommends adding the following dialing patterns:

  • 1NXXNXXXXXX

  • N11

  • N33

  • 011 (The PBX needs to strip the 011)

 

7) Set the "Trunk Sequence" for "Matched Routes":

  • xxxxxxxxxxGW1

  • xxxxxxxxxxGW2

 

IMPORTANT: Ensure that you send a legitimate 10 or 11-digit caller ID. For more information, see Setting Up Caller ID

 

If you experience issues with your configuration after following this guide, please contact the Support team at support@siptrunk.com.

Was this article helpful?
2 out of 5 found this helpful
Have more questions? Submit a request

Comments

  • Avatar
    Roddy Dairion

    What is the dialplan suppose to look like? Mine look like this

    exten => _.,1,AGI(agi://127.0.0.1:4577/call_log)
    exten => _.,n,Dial(${SIPTRUNKA}/${EXTEN},,tTor)
    exten => _.,n,Hangup

    Yet I can't make calls to Mauritius for example? Is this normal?

  • Avatar
    Nitesh Katoch

    Try this

    exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
    exten => _X.,n,Dial(sip/${EXTEN}@SIPTRUNK,55,to)
    exten => _X.,n,Hangup

Powered by Zendesk