skype and asterisk
Created by: telecomchina,Last modification on Sat 27 of Oct, 2007 [15:48 UTC] by linkx
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- Re: kamailio vs opensipsby bogdan_iancu Thu 20 of Nov, 2008Of course a web site cannot provide a comprehensive view over a project. But bottom line, after OpenSER split, the developers for most critical parts (like TM, core, presence, dialog, mediaproxy) went for OpenSIPS - a project (as power to delivered) is defined by the active and skilful developers. Kamailio recently announce merging with SER in order to go back and use the core and TM of SER (probably because of lack of devs) - anyhow, a merging back to SER will invalidate everything that made OpenSER so successful and apart from SER.
OpenSIPS has ongoing work for new a lot of new adds-on - more dialog stuff, prepaid controller, RLS, B2BUA, QoS, advanced routing, real load-based balancing, etc - maybe (as most of the tech guys) the listing of all this on the project web page is not the best (tech guys speak code not words :D).
... - Re: DNS Configby lotsofish Thu 20 of Nov, 2008Add an A record to the nameserver for your domain. For example, if you have a domain called example.com, add an A record to the nameserver records for "asterisk" and the ip of your external IP.
Then you can connect to your asterisk server using asterisk.example.com - Re: kamailio vs opensipsby asiby Wed 19 of Nov, 2008Ask them both the same question and choose the one from which you will received the quickest response. Cause you don't want do deal with potentially almost dead projects like SER. Stick with the one that will have the biggest dev team and the most active community.
Cheers
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