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Asterisk is an extremely powerful piece of open source software that gives you the ability to run a full-featured software based PBX on your computer.

Asterisk is NOT plug and play software and because of its extremely versatile nature, it can be difficult for first-time users to setup.

Below is a basic setup guide, as well as links to several resources that you can use to assist you in configuring your Asterisk system. Due to the complexity and dynamic nature of the Asterisk Platform, unfortunately we cannot assist in fully customizing your install.

DID based routing with Asterisk

Please note, Callcentric is not responsible for preventing unwanted physical or remote access to your IP PBX. If your IP PBX is compromised, you will be responsible for all associated damages.

Please be sure to read this guide regarding securing your IP PBX solution.

RESOURCES

Websites
Asterisk Homepage

Help / Support:
Asterisk Support Page
Asterisk Forum
Asterisk Wiki
Broadband Reports VoIP Forum

Configuring Asterisk 17 - (chan_pjsip)

The instructions below are meant to assist you with the basic configuration of Asterisk (PJSIP). We recommend reading each step through in its entirety before performing the action(s) indicated within the step.

We also recommend checking which version of Asterisk your PBX is based on, as there are significant differences between each revision. To check which version your PBX is based on, please log into your PBX's command line interface and execute the command core show version. In doing so, you should see an output similar to the following:

Asterisk 17.5.1 built by root @ server on a x86_64 running Linux on 2020-06-19 22:40:24 UTCC

STEP 1	Setting up your trunk and global options
	Edit the pjsip.conf file with your favorite text editor and make the following changes: Add the following underneath the [global] section of your pjsip.conf file: [transport-udp] type=transport protocol=udp bind=0.0.0.0:5060 Underneath that, add the following section to define the Callcentric trunk/peer: ; ----------------------------------- ; CALLCENTRIC TRUNK ; ----------------------------------- [callcentric] type=registration transport=transport-udp outbound_auth=callcentric_auth retry_interval=60 expiration=3600 auth_rejection_permanent=yes contact_user=1777MYCCID server_uri=sip:sip.callcentric.net client_uri=sip:[email protected] [callcentric_auth] type=auth auth_type=userpass password=SUPERSECRET username=1777MYCCID [callcentric] type=endpoint transport=transport-udp context=from-callcentric disallow=all allow=ulaw outbound_auth=callcentric_auth aors=callcentric from_domain=sip.callcentric.net from_user=1777MYCCID sdp_owner=1777MYCCID direct_media=no ice_support=no send_rpid=yes rtp_symmetric=yes force_rport=yes timers=no [callcentric] type=aor contact=sip:[email protected] [callcentric1] type=identify endpoint=callcentric match=204.11.192.0/22 [callcentric2] type=identify endpoint=callcentric match=199.87.144.0/21 NOTE: Any instance of 1777MYCCID should be replaced with the 1777 number assigned to you by Callcentric. EXT is the three digit extension you are trying to register this UA to. For example, 17770001234101 would register to "extension 101" on account "17770001234". However, if you are registering towards the main/default extension of your Callcentric Account (extension 100), you can simply use your 1777 number without the 3 digit extension suffix (example: 17770001234). You CANNOT register to your Callcentric Account by only using the extension number; the full 1777 number ALWAYS needs to be included. Also be sure to replace SUPERSECRET with your Extension's SIP Password. Your Extension's SIP Password is the password you created for the extension you are trying to use. You can edit your Extension's SIP Password by logging into your Callcentric Account and clicking on the Extension menu link and then selecting the extension you would like to update. Add an extension to handle calls to/from your SIP phone. We've detailed a sample below (100 and 101 are extensions of your phones); for further information please refer to the documentation for both Asterisk and your SIP Phone. ; ======================== ; EXTENSIONS ; ---------------------- ; Extension 100 ; ---------------------- [100] type=endpoint aors=100 auth=100-auth allow=ulaw context=to-callcentric callerid="First Extension" <100> ice_support=no rtp_symmetric=yes force_rport=yes rewrite_contact=yes dtmf_mode=rfc4733 [100-auth] type=auth auth_type=userpass password=PASSWORD username=100 [100] type=aor max_contacts=1 remove_existing=yes qualify_frequency=60 ; ---------------------- ; Extension 101 ; ---------------------- [101] type=endpoint aors=101 auth=101-auth allow=ulaw context=to-callcentric callerid="Second Extension" <101> ice_support=no rtp_symmetric=yes force_rport=yes rewrite_contact=yes dtmf_mode=rfc4733 [101-auth] type=auth auth_type=userpass password=PASSWORD username=101 [101] type=aor max_contacts=1 remove_existing=yes qualify_frequency=60 NOTE: Replace any instances of PASSWORD with the desired password for each individual extension. Once completed, please make sure to save your changes.
STEP 2	Configuring call handling
	Edit the extensions.conf file with your favorite text editor and make the following changes: Add the following section to route calls FROM Callcentric TO your SIP Phone with extension 100: [from-callcentric] exten => 1777MYCCID,1,Dial(PJSIP/100) Alternatively, if you've acquired DIDs from us and would like to perform a DID Based Routing Solution (route inbound calls towards a specific destination based on the number that was dialed) you can use the following: [from-callcentric] exten => 1777MYCCID,1,Set(Var_TO=${CUT(CUT(PJSIP_HEADER(read,To),@,1),:,2)}) exten => 1777MYCCID,2,GotoIf($["${Var_TO}" = "1777MYCCID"]?to-100,s,1:3) exten => 1777MYCCID,3,GotoIf($["${Var_TO}" = "DID1"]?to-100,s,1:4) exten => 1777MYCCID,4,GotoIf($["${Var_TO}" = "DID2"]?to-101,s,1:5) exten => 1777MYCCID,5,Hangup [to-100] exten => s,1,Dial(PJSIP/100,20) exten => s,n,Hangup [to-101] exten => s,1,Dial(PJSIP/101,20) exten => s,n,Hangup NOTE: Any instance of 1777MYCCID should be replaced with the 1777 number assigned to you by Callcentric. EXT is the three digit extension you are trying to register this UA to. For example, 17770001234101 would register to extension 101 on account 17770001234. However, if you are registering towards the main/default extension of your Callcentric Account (extension 100), you can simply use your 1777 number without the 3 digit extension suffix (example: 17770001234). You CANNOT register to your Callcentric Account by only using the extension number; the full 1777 number ALWAYS needs to be included. Where DID1, and DID2 are located, please replace it with the number(s) you've acquired from our service. For example, if you've acquired the number 12125551000, the following line: exten => 1777MYCCID,3,GotoIf($["${Var_TO}" = "DID1"]?to-100,s,1:4) Would look something like: exten => 17770000001,3,GotoIf($["${Var_TO}" = "12125551000"]?to-100,s,1:4) NOTE: To explain the context above, as we will always send the phone number that is being dialed in the TO header of the SIP INVITE packet; the context will extract the DID from the TO header of the SIP INVITE packet and then perform a comparison. From the example above, if DID1 was extracted, it will send the call to the context labeled to-100, which then sends the call to extension 100. Add the following section to route calls FROM your SIP phone TO Callcentric: [to-callcentric] exten => _X.,1,Dial(PJSIP/${EXTEN}@callcentric) Once completed, please make sure to save your changes. If necessary restart (or start) the asterisk server using the proper command: /etc/init.d/asterisk restart, restart asterisk, service asterisk restart... etc.
STEP 3	Connect to your Asterisk PBX and verify connections
	To connect to your PBX System, use the IP Address (or hostname) for your Asterisk Server in conjunction with "100" (the extension created earlier which is the username) and the password for the "100 extension". Connect to the asterisk console by running the following from the command line: `asterisk -r` Verify that Asterisk is registered to Callcentric with the console command pjsip show registration `CLI> pjsip show registrations <Registration/ServerURI..............................> <Auth..........> <Status.......> ========================================================================================= callcentric/sip:sip.callcentric.net callcentric Registered` Verify that your SIP phone is registered to Asterisk with the console command pjsip show endpoints* Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.> I/OAuth: <AuthId/UserName...........................................................> Aor: <Aor............................................> <MaxContact> Contact: <Aor/ContactUri...............................> <Status....> <RTT(ms)..> Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................> Identify: <Identify/Endpoint.........................................................> Match: <ip/cidr.........................> Channel: <ChannelId......................................> <State.....> <Time(sec)> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......> ========================================================================================= Endpoint: 100/100 Not in use 0 of inf InAuth: 100-auth/100 Aor: 100 1 Contact: 100/sip:[email protected]:1030;line=jybz6e1w Avail 22.749 Identify: 100-identify/100 Endpoint: 101/101 Not in use 0 of inf InAuth: 101-auth/101 Aor: 101 1 Contact: 101/sip:[email protected]:5260 Avail 7.933 Identify: 101-identify/101 If your output is not similar to that which is detailed above, then your SIP phone is not configured correctly. Disconnect from Asterisk by typing exit. Asterisk will remain running in this case.
STEP 4	That's it! You can now make a phone call
	You can make a test call to 17771234567, or if you are signed up for one of Callcentric's rate plans you can place a call to a traditional landline or mobile phone by dialing either: 1 + the area code and number for calls to the US Or 011 + the country code, area code, and number for calls worldwide (you may also use 00 instead of 011).