Hi Veselin,
You can verify SDP and RTP by running protocol analyzer such Ethereal, if
you need instruction you can follow this link.
http://www2.cs.uh.edu/~jsteach/cosc4377/2000fall/ethereal.html
While TDM side, I think he is referring to the card, if the card is faulty
or not.
Best Regards,
Do you mean only once per day?
On Dec 1, 2007 7:47 AM, Alex Balashov [EMAIL PROTECTED] wrote:
Store a value indicating it has been called as a unique key in AstDB, and
set your dial plan to check for it.
On Fri, 30 Nov 2007, [EMAIL PROTECTED] wrote:
Anyone have an idea how to implement a
Hi Jake,
Perhaps you can add
NoOp(${PLAYBACKSTATUS}) after each Playback, it should return either
FAILED or SUCCESS in the CLI
Hope that helps.
Best Regards,
Joanna
On 2/28/07, Kuba [EMAIL PROTECTED] wrote:
After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very
strange
Hi Cameron,
Why not automatically set the language that should be use at the beginning.
Set(LANGUAGE()=nz)
Hope that helps.
Best Regards,
Joanna
On 2/28/07, Moises Silva [EMAIL PROTECTED] wrote:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetLanguage
There you can found how
Hi Al,
We have different email contents and subject depending in the action that
the customer had taken, since our service is a Prepaid card. What I did, was
to use an AGI to have this flexibility. After getting the caller id and
querying it in the DB, I will then construct the email that will
,
Joanna Liza Mariazeta
www.mariazeta.com
On 2/21/07, Jon Farmer [EMAIL PROTECTED] wrote:
Hi
I am writing a IVR app using phpagi and are coming up against a problem
when trying to detect DTMF. If I use the get_data function I dont seem to be
able to reliably detect 16 digits. If I try 10 digits
Hi Pierre,
Just a thought..check your dtmfmode in your SIP client configuration, if
your using inband but your codec is not ulaw or alaw the DTMF tones will be
misrepresented and thus will not be recognised due to the audio compression,
on the other hand if your phones are rfc2833 and asterisk
Hi there,
Why not use the command Set, and set the variable exten =
558,3,Set(CALLERID(number)=111)
Hope that helps,
Best Regards,
Joanna Liza Mariazeta
www.mariazeta.com
On 2/20/07, Mohamed Farid [EMAIL PROTECTED] wrote:
Dear All :
I need to mask the caller ID and pretend to make
Hi Pierre,
You can also add the following if you think its helpful.
relaxdtmf = yes ;relaxes the DTMF detection parameters
qualify = yes ; will send a SIP Optoin command regularly to check if the
device is still online, if the device did not answer within 2 seconds it
will be considered
can choose language
(English,Mandarin). And then they will be passed to the queues. Take note of
the Queue(queue-out-English|tThH|||60), the option T allows the calling user
to transfer the call when they pressed a single digit, while in the queue.
Hope that helps.
Best Regards,
Joanna Liza
-rows();
if ($ret 0) {
$AGI-set_variable('ISREGISTERED', '1');
exit;
} else {
$AGI-set_variable('ISREGISTERED','0');
}
Hope that helps..
Best Regards,
Joanna Liza Mariazeta
www.mariazeta.com
On 2/8/07, prasanth [EMAIL PROTECTED
Regards,
Joanna Liza Mariazeta
www.mariazeta.com
On 2/15/07, Jason Wolfe [EMAIL PROTECTED] wrote:
Hello,
I have the following simple application...
1. Call is answered, and Dial() function is used with a macro to dial
out to a number.
2. 'Called' party answers the phone, and hears a message
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