http://www.washington.edu/computing/mailman/faqs/mailman.email.html
Em 07/01/2010, às 15:29, C. Chad Wallace
cwall...@lodgingcompany.com escreveu:
At 2:01 PM on 07 Jan 2010, Dan Journo wrote:
I've never seen that in Outlook. What client do you use?
Claws Mail provides a Mailing-List
I think that's very wise advice. To offer a commercial perspective, our
customers willing to pay for sophisticated
smart phone apps (currently gov/mil agencies and some mid-size telecoms)
have very specific needs and care about reliable operation, development
duration, long-term support --
Jailbreak your iPhone and install Cydia to have a Unix like open
source environment (based on Debian), then install Siphon SIP client,
and have fun!
Regards.
Em 05/01/2010, às 18:04, UIT DEVELOPMENT uit...@gmail.com escreveu:
Yep. Its called unemployment. Got the iPhone a little less
Store.
Regards.
Em 06/01/2010, às 03:21, Randy R randulo2...@gmail.com escreveu:
On Wed, Jan 6, 2010 at 6:48 AM, Allann Jones allan...@gmail.com
wrote:
Jailbreak your iPhone and install Cydia to have a Unix like open
source environment (based on Debian), then install Siphon SIP client
Modify the IRQ used by device on BIOS. Rearrange IRQs until the IRQ is
used only by the card or it is shared with a unused device. Some
device drivers can have IRQ hard-coded on source code.
Em 03/01/2010, às 01:18, Greg Woods g...@gregandeva.net escreveu:
On Sat, 2010-01-02 at 20:25
http://www.voip-info.org/wiki-Asterisk+variables
http://www.voip-info.org/wiki-Asterisk+Dialplan+Globals
Beware with the variable scope, if its scope is local the values can
be lost, so declare the variable as global when its value must persist
between calls.
[globals]
FOOBAR = NO
Hi. I'm writing a speech recognition module for Asterisk. I'm having
problems with simultaneous SIP and PSTN calls. Sometimes Asterisk crashes in
this scenario. I don't have problem with simultaneous calls using PSTN calls
only.
The implementation is in the file res/res_speech.c
Does someone know
Hi. Are the audio streams returned by the user been shared between SIP and
PSTN connections? I'm developing a speech recognition engine for Asterisk
and I'm facing a problem where Asterisk is crashing when concurrent SIP and
PSTN connections occur. I will read the code that implement that to
Internal and external calls can be distinguished generally by the phone
number. A prefix or the number of digits of the phone number. For example,
you could use a digit prefix followed by a interval of time to call a
internal number.
Examples:
Internal number: 0,1234
External number:
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
On Tue, Jul 15, 2008 at 2:37 PM, Fidel Garcia [EMAIL PROTECTED]
wrote:
This one!
The sound of a phone that signals a call coming from internal/external
My phones are SIP, I do not know what ZAP means or what it does.
Thanks for your
I'm with a problem. When a timeout occurs in a WaitExten() function call,
the 'h' extension is taking the priority and the 't' extension is not
executed. The 't' extension is executed only when I delete the 'h' extension
entry.
exten = s,n,WaitExten()
exten = t,1,Goto(from-pstn,s,4)
exten =
Hi. I'm writing a speech recognition engine for Asterisk. I'm having a
problem with ast_speech_write, the audio data is coming with a silence of
0.20ms at each between 0.20ms of normal audio. The audio data configured in
ast_speech_new is AST_FORMAT_SLINEAR. Am I wasting some needed
configuration?
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