I’m posting here as I find the AstLinux community to be the most friendly and knowledgeable about all things Asterisk!
My ITSP offers DID/SIP trunks at a very competitive rate–each DID includes 5 channels. I’m running Asterisk 11. The ITSP only offers a single server for both incoming and outgoing calls: sip05.unlimitel.ca. I recently discovered that with the configuration I had, all of my calls, no matter how many DIDs I have, were being sent over a single trunk (the first to register from Asterisk). I believe this is due to my selecting type=peer in my SIP.conf; it appears to match based on IP & port so all of the DID/trunks appear as a single one. This means that I’m limited to 5 channels, despite having 4xDIDs which should give me 20 channels (5 per DID). I’ve tried changing this to type=user, but although I see registration (sip show registry) and users (sip show users) I cannot see any incoming calls. Can anyone offer any help/suggestions? Pulling my hair out! Shamus ------------------------------------------------------------------------------ The best possible search technologies are now affordable for all companies. Download your FREE open source Enterprise Search Engine today! Our experts will assist you in its installation for $59/mo, no commitment. Test it for FREE on our Cloud platform anytime! http://pubads.g.doubleclick.net/gampad/clk?id=145328191&iu=/4140/ostg.clktrk _______________________________________________ Astlinux-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to [email protected].
