Hello,
Thank you for your reply.
I should indeed have been more specific :-)
For the moment, we plan to implement about 10 lines with 2 "head numbers" that will be used to rotate calls depending on the hours and such.
Your suggestion to implement our own system is a valid one but is not one we would like to pursue. I have no doubt there are very good open source solutions but given the reasonable price of the solutions that we have been seeing, it does not seem like it would make sense to set up our own system.
Regarding UPC Cablecom, I have not been successful at reaching them (maybe because our planned setup is so small ?) but I should probably give it another try.
Have a nice day,
On Aug 29, 2014, at 8:45 AM, Jeroen Massar jeroen@massar.ch wrote:
On 2014-08-29 03:23, Michael Horn wrote:
On Thu, 28 Aug 2014 19:45:43 +0200 (CEST) Ralph Krämer ralph.kraemer@vable.ch wrote:
I suggest to run your own Asterisk (or Callmanager)
...or freeswitch, if you like reliability and predictable behaviour.
I've managed quite a few freeswitch deployments until now and did some migrations from Asterisk to freeswitch due to issues with reliability or flexibility. As an office/corporate pbx I'd select freeswitch over asterisk any time. If something more specialized is needed, yate is worth a try.
+1 on FreeSWITCH, have been using that for years; and as they finally have debian packages that work in their own apt-repo, makes installing much easier too.
One thing to note though is that one does not have to keep their configfile layout, one can bend it anyway one likes.
through a redundant SIP-Trunk to an VoIP Provider.
Curious however about recommendations for VoIP providers in .ch there aren't that many around unfortunately. would love to hear some suggestions (backed by experiece).
I have good experience with netvoip.ch; though it really depends on why/how one uses it (in my case inbound only mostly and not for large volumes).
Note also, that if one has a hardline and a Fritz!Box, one can use the latters internal SIP server to make that hardline available from anything you can connect to it. (SIPclients -> FS -> F!B in my case).
Greets, Jeroen
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