On Tue, 24 Jun 2008 08:28:35 +0200, randulo wrote: >On Tue, Jun 24, 2008 at 5:58 AM, C. Savinovich ><[EMAIL PROTECTED]> wrote: >> >> To be fair, Centile is better geared than asterisk for virtual pbx >> hosting. It comes with a system to manage virtual pbxs... it also handles >> the provisioning of most ip phones adequately, it is a totally different pbx >> although linux based. > >Interesting. Yes, it has a few phones it knows how to provision. I am >using "generic SIP device" for both the phones currently in use. > >>Although I don't know the details of your setup, it >> would not surprise me to see Centile accepting 2 different phones with the >> same extension on the same pbx. > >Well, my 4AM brainstorm didn't help. The phone I'm having trouble with >is my favorite one, a Siemens S675IP. It is registered and works >perfectly with 5 other SIP providers. On the Centile pbx, it can make >calls but it can not be called. The web admin interface shows the >correct public and NAT ip addresses and shows the phone "in service". >Calling it from another phone rings once and then goes to congestion, >or at least that's the signal I hear. (It's wierd not being able to >ssh in and see what's happening.)
Randy, This is exactly what was happening when I used an Aastra 480i CT with OnSIP. According to OnSIP it's not a supported phone, although the newer 57i CT does work with OnSIP. It seemed that the phone was losing registration with the provider. I was not able to overcome this in the phone or provider settings. My ultimate solution was to build a small Asterisk instance (Astlinux) on a thin client (HP T5700) and use it strickly as a bridge device for the phone. For whatever reason, the Astlinux box could sustain the registration and pass the incomming calls to the phone. This is very similar to another idea that I once had but never actually implemented. That is, using a small embedded Asterisk device as a SIP<>IAX2 protocol translator to facilitate complex NAT traversal. I thought that Astlinux on Gumstix hardware would be ideal for such a task. Michael -- Michael Graves mgraves<at>mstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
