[asterisk-dev] Start testing with res_sip

[Olle E. Johansson]

19 jun 2013 kl. 16:11 skrev Malcolm Davenport <malcolmd at digium.com>:

> The documentation is actually contained in the res_sip source file itself.

More than 10 years ago I got answers like this from developers, which lead to me filling James' voip-info.org site with the docs from the source code and then to me starting to develop myself. :-)
But this is a developer mailing list, so that answer is fine here... Just a flashback from an old man.

I thought the channel driver is called chan_pjsip - not res_sip?

/O
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