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 <sasa.bobek...@gmail.com>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 < > carlos.ruizd...@gmail.com> 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 <sasa.bobek...@gmail.com>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 < >>> carlos.ruizd...@gmail.com> 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)
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