I’m using Cisco Linksys SPA3102 to connect to asterisk. I have followed the official user manual and the blog post here http://www.skelleton.net/2012/08/02/linksys-spa-3102/
Note: The Linksys SPA3102 provides the ability to connect standard telephones and fax machines to IP-based data works with the additional benefit of an integrated connection for legacy telephone work “hop-on hop-off” applications.
When I call an extension say 225 from the analog phone, I can get the IVR I have setup in my dialplan. But when I Call the analog phone extension using a sip phone I get the following error message:
“Unable to create channel of type ‘SIP’ (cause 20 – Subscriber absent)” which usually means that the SIP device hasn’t registered yet. If you are having RTP problems, get sure that uuid-devel and libuuid-devel are installed.
Also note that type=friend should normally be type=peer in this context and for security reasons, it is not advisable to use an extension number as a SIP resource name.
You might also want to take a look at the following guide: SIP Demystified, which tells you why the standard is needed, what architectures it supports, and how it interacts with other protocols. As a bonus, you even get a context-setting background in data networking. Perfect if you’re moving from switched voice into a data networking environment, here’s everything you need to understand:
- Where, why, and how SIP is used
- What SIP can do and deliver
- SIP’s fit with other standards and systems
- How to plan implementations of SIP-enabled services
- How to size up and choose from available SIP products