@Steve: I considered the hardware separation between servers but when I exposed the idea it was immediately discarded because it is mandatory to have all in a box.
Well, I'll start the migration then. Thank you. On Sun, Jul 19, 2009 at 12:59 AM, Steve Totaro < [email protected]> wrote: > No need to migrate, just have a chan_mobile server to hand the calls over > via SIP. > > It is your "cell phone network gateway" > > I like to separate functions to different boxen. Database on one, Asterisk > on another, TDM <-> SIP gateway on another, GUI/CRM somewhere else. Why not > have a Cell <-> SIP gateway? > > Just my approach but it seems to work well. Power and RU space aside. > > Thanks, > Steve Totaro > > > On Sat, Jul 18, 2009 at 11:23 PM, Sasa Bobek <[email protected]>wrote: > >> Yes, chan_mobile works great on Elastix. If the migration is complicated, >> you may consider installing/testing it on an old computer. >> >> >> On Sun, Jul 19, 2009 at 2:21 AM, Carlos Ruiz Diaz < >> [email protected]> wrote: >> >>> Thank for your time. >>> >>> Do you used chan_mobile with Elastix distribution successfully? If so, I >>> will consider the switch. I can't jump to another distribution easily >>> because I have a working environment that will make really hard the >>> migration. >>> >>> >>> On Sat, Jul 18, 2009 at 10:57 AM, Sasa Bobek <[email protected]>wrote: >>> >>>> In general, I found it hard to get chan_mobile working straight out of >>>> the box, and although there is a great effort to make it so, phone >>>> manufacturers are not helping by making command sets and BT implementations >>>> different from device to device, SW version to SW version. Elastix seems >>>> to >>>> have the most trouble free implementation out there and has certainly saved >>>> me a lot of time and money and I recommend you give it a go, before banging >>>> your head over code. You can check the buglist on Digium for further info >>>> or the list of compatible phones on voip-info.org, but it may be a USB >>>> dongle issue as well (CSR seems to be the safest bet after they fixed the >>>> error log flood). >>>> >>>> On Sat, Jul 18, 2009 at 3:27 PM, Carlos Ruiz Diaz < >>>> [email protected]> wrote: >>>> >>>>> Hello >>>>> >>>>> I recently updated my asterisk-addons-1.6.2 to the last revision and I >>>>> have this problem that I don't know how to interpret, bug or not. I >>>>> connected a Nokia N80 phone to use chan_mobile and everything works great >>>>> until the phone starts getting disconnected after the call finished and >>>>> sometimes during the call attempt. >>>>> >>>>> Is this a bug or a possible known issue for Nokia phones? >>>>> >>>>> # rpm -qa | grep blue >>>>> >>>>> pulseaudio-module-bluetooth-0.9.12-10.1 >>>>> bluez-utils-3.36-7.1 >>>>> kdebluetooth4-0.3-4.1.1 >>>>> libbluetooth-devel-3.36-3.1 >>>>> gnome-bluetooth-0.11.0-26.2 >>>>> bluez-test-4.22-6.1.1 >>>>> libbluetooth3-4.22-6.1.1 >>>>> libbluetooth2-3.36-3.1 >>>>> >>>>> Thanks in advance! >>>>> >>>>> Carlos. >>>>> >>>>> _______________________________________________ >>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>>> >>>>> asterisk-users mailing list >>>>> To UNSUBSCRIBE or update options visit: >>>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>>> >>>> >>>> >>>> _______________________________________________ >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>> >>>> asterisk-users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>> >>> >>> _______________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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