[asterisk-users] Centile ipbx, anyone heard of this?
Michael Graves
mgraves at mstvp.com
Tue Jun 24 09:50:45 CDT 2008
On Tue, 24 Jun 2008 08:28:35 +0200, randulo wrote:
>On Tue, Jun 24, 2008 at 5:58 AM, C. Savinovich
><c.savinovich at itntelecom.com> 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:mjgraves at pixelpower.onsip.com
skype mjgraves
54245 at fwd.pulver.com
More information about the asterisk-users
mailing list