Displaying 6 results from an estimated 6 matches for "virsions".
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2005 Jul 27
2
oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6
Abwesenheitsnotiz: [Asterisk-Users] oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6Hi All
I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb
ram, with g729 for i686 , (fedora 1).
my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen
otherparty realtime voice , but other party geting sip party's voice 1 sec
later (not
PG gem behaves strange. It requires live DB connection to only generate a model. How to turn it off?
2013 May 01
3
PG gem behaves strange. It requires live DB connection to only generate a model. How to turn it off?
I used MySQL as a backend before. Now I had to move to PG since Heroku
doesn''t have MySQL and forces everyone to use their Heroku-PG database.
But PG virsion of Rails works diffidently. I cannot generate a model
without PG server running:
>> C:\ruby\Heroku\App>rails generate model Products
>>
>> C:/RailsInstaller/Ruby1.9.3/lib/ruby/gems/1.9.1/gems/activerecord-
2012 Mar 31
20
Need help with a windows app
so i was trying to run a game and when i ran it via terminal i got some answers why.
fixme:ntdll:NtLockFile I/O completion on lock not implemented yet
fixme:keyboard:X11DRV_LoadKeyboardLayout L"00000409", 0080: stub!
fixme:keyboard:X11DRV_LoadKeyboardLayout L"00000409", 0001: stub!
fixme:wgl:X11DRV_wglChoosePixelFormatARB unused pfAttribFList
wine: Unhandled exception
2005 Jan 06
1
calling with out registration
hi,
i am using Asterisk CVS-05/31/04.
i have the problem that sip clients can make calls over asterisk
without registering befor. the xlite is not loged in with any
username/secret bit still can make calls over asterisk.
how can that be?
thx for help.
thomas
2001 Feb 10
3
Protocol 2 remote forwarding patch
Hi all,
I'm very new in this list, as looking for codes to plug up the lack of
functionality of "Protocol 2 Remote Forwardig".
Fortunately, I could find it in MARC's archive. Mr. Jarno Huuskonen
posted the codes in Sept, last year, and I tried applying it to my
FreeBSD box environment.
I couldn't apply an original patch, of course, for incompatibility of
virsion. The
2005 Jul 27
2
Regarding Call Hold
> Hi All,
>
> We are using asterisk for testing our home gateway setup.
> We have implemented Call Hold feature in our application.
> In our Application we have written code in such a way that for an INVITE
> for
> putting a SIP phone on HOLD will contain connection address "0.0.0.0" in
> the SDP message.
> We expect the same connection address i.e