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Documentation Index

Fetch the complete documentation index at: https://docs.getsmartalex.com/llms.txt

Use this file to discover all available pages before exploring further.

Audience: Asterisk admin comfortable editing pjsip.conf. Time: 10 minutes. Driver: chan_pjsip.

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Prerequisites

  • Asterisk 16 or later (17, 18, 20, 21 all work)
  • chan_pjsip loaded (pjsip show version)
  • Outbound UDP 5060 allowed
  • SIP ALG disabled upstream

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Step 1: create trunk in SmartAlex

SIP Trunk wizard, pick Asterisk. Save and copy credentials.

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Step 2: pjsip.conf configuration

Append to /etc/asterisk/pjsip.conf (or use a separate conf file): 
[smartalex-transport]
type=transport
protocol=udp
bind=0.0.0.0

[smartalex-registration]
type=registration
outbound_auth=smartalex-auth
server_uri=sip:sip.voice.getsmartalex.com
client_uri=sip:YOUR_USERNAME@sip.voice.getsmartalex.com
retry_interval=60
expiration=3600

[smartalex-auth]
type=auth
auth_type=userpass
username=YOUR_USERNAME
password=YOUR_PASSWORD

[smartalex]
type=aor
contact=sip:sip.voice.getsmartalex.com

[smartalex]
type=endpoint
transport=smartalex-transport
outbound_auth=smartalex-auth
aors=smartalex
context=from-smartalex
disallow=all
allow=ulaw,alaw
direct_media=no
from_domain=sip.voice.getsmartalex.com
rewrite_contact=yes

[smartalex]
type=identify
endpoint=smartalex
match=sip.voice.getsmartalex.com
Replace YOUR_USERNAME and YOUR_PASSWORD with values from SmartAlex.

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Step 3: extensions.conf dialplan

Add a context for inbound calls from SmartAlex: 
[from-smartalex]
exten => _X.,1,NoOp(Inbound from SmartAlex: ${EXTEN})
exten => _X.,n,Goto(from-internal,${EXTEN},1)
Reload: 
asterisk -rx "pjsip reload"
asterisk -rx "dialplan reload"

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Step 4: verify registration

asterisk -rx "pjsip show registrations"
Expected: smartalex showing Registered.

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Step 5: add extensions in SmartAlex

Settings, then PBX. Populate the directory. Agent Studio, then Telephony tab. Link the trunk. Call, verify, transfer.

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Asterisk-specific notes

  • SIP tracing: asterisk -rvvvv then pjsip set logger on.
  • TLS transport: add a type=transport section with protocol=tls and point to your cert bundle. Use port 5061.
  • Custom headers: use PJSIP_HEADER() in dialplan if you need to inspect P-Asserted-Identity or other SIP headers on transferred calls.
  • Multiple SmartAlex trunks: each trunk needs its own unique section names (append -2, -3, etc.).

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Next steps

FreePBX

GUI on top of Asterisk.

Troubleshooting

Error reference.