T11 · T11-R02
Asterisk and FreeSWITCH as the other PBX
Learn it in 5 minutes
- Wikipedia describes Asterisk as a software PBX that connects phones, the PSTN, and VoIP endpoints.
- The Asterisk documentation opened here separates module types. Channel drivers are one type. res_pjsip is the SIP channel configuration the same docs link as Configuring res_pjsip.
- Wikipedia describes FreeSWITCH as telephony software for real-time audio, video, and text, including SIP and WebRTC.
- The FreeSWITCH users manual that opened is the operator book. A Sofia SIP chapter URL redirected to that manual, so this lab links the manual.
- Neither product copies a CUCM calling search space. You translate the idea, then you test one call.
Visual
What is going on
Asterisk matches calls in a dialplan. FreeSWITCH matches calls in its own dialplan. CUCM matches a partition and a route pattern. The interop skill is to name the digit string on each side of the trunk and the codec each side offered. The Asterisk module page shows channel-driver modules as a category. It does not list your peer's host. The FreeSWITCH manual says its parameters are checked against the project source and the shipped defaults. When a guide and a blog disagree, the guide for the version you installed is the one to follow.
Confirm before relying. Confirm the module list and the SIP port in the version you installed. This lab does not copy a sample peer.
Worked example
Asterisk answers a lab INVITE with 488 Not Acceptable Here. CUCM offered a codec the peer's endpoint did not allow. Add a codec both sides list, or remove the one they cannot use. Do not restart CUCM to test a codec the offer already showed.
Quiz
1. An Asterisk dialplan is a CUCM calling search space.
False. It is a different object on a different product.
Ecosystem
Sister pieces sit on the ecosystem page: the glossary, the mind map, the integration map, and the sibling labs.
- Dialplan — The Asterisk or FreeSWITCH rules that match a call. Not a CUCM CSS.
- res_pjsip — The Asterisk SIP channel stack named in the documentation opened for this lab.
Config. Record the version string of the PBX and the one match that should take the test call.
Troubleshooting. A 401 from the far PBX is authentication. A 404 is a miss in that dialplan. Read the status before you change CUCM.
Bridge. CUBE can sit between them and still be only a B2BUA. Two ladders remain two ladders.
Mind map, sister technologies, and future hooks live on the ecosystem page.
Sources
- Asterisk (Wikipedia)
- Types of Asterisk modules
- Configuring res_pjsip
- FreeSWITCH (Wikipedia)
- FreeSWITCH users manual
Product screens move. Confirm the guide for your release before you change a live system.
Sources and go deeper
Opened 2026-10. A row marked confirm has no link, because that page was not verified.
- Wikipedia. Asterisk (Wikipedia) last checked 2026-10
- Asterisk. Types of Asterisk modules last checked 2026-10
- Asterisk. Configuring res_pjsip last checked 2026-10
- Wikipedia. FreeSWITCH (Wikipedia) last checked 2026-10
- FreeSWITCH. FreeSWITCH users manual last checked 2026-10
- RFC. RFC 3261, SIP last checked 2026-10