So, after some research I can provide you with some more information.
According to our employees on every fourth call the dialtone is choppy.
That happens, not like I said first, when we dial trough phpagi AND when
we dial directly with x-pro (but both times through asterisk).
In X-Pro its a bit better, but still a bit choppy.
The other problem is that the whole call is choppy.
What actually could be a bandwith problem... (although it's mysterious
because the first 2-3 weeks that problem hasn't occured (while approx. 5
people worked like today (2005-09-29))
Another problem is that asterisk hangs up the call sometimes after some
seconds without interaction of the user. I don't know why :-/
So I think one part of the solution is a higher bandwidth.
Butthe hangup-problem is another kind of problem I think.
Do you have some ideas what could causes such problems (as described above)
Regards
Michael
Michael Häberle wrote:
Ok :)
the dialplan looks like that (mynumber is a tel-number):
-------------------------------------------------
[general]
static=yes
writeprotect=no
[telout]
exten => _X.,hint,SIP/41
exten => _X.,1,dial(SIP/${EXTEN})
exten => _X.,2,SetCIDName(anonymous)
exten => _X.,3,dial(SIP/[EMAIL PROTECTED],30,r)
exten => _X.,4,Hangup
-------------------------------------------------
I dial out of a webapplication, when I press a button, we connect to
asterisk through phpagi.
here are the php-functions:
function startCall($number,$uid) {
$returnValue = false;
$state = getStatus();
if ($state >= 0 && $state <4) {
$asm = new AGI_AsteriskManager();
if($asm->connect())
{
$call = initCall($asm, $number);
$asm->disconnect();
if (trim($call['Response']) == "Error") {
$returnValue = false;
} else {
$returnValue = true;
}
} else {
echo "Connect to Asterisk FAILED";
}
} else {
echo "Extension already in use";
}
function initCall($asm, $number) {
$call = $asm->send_request('Originate',
array('Channel'=>"SIP/" . $_COOKIE['extension'],
'Context'=>'telout',
'Exten'=>$number,
'Priority'=>1,
'Timeout'=>30000,
'Async'=>false,
'Callerid'=>'anonymous'));
return $call;
}
for the cookie we have defined a channel in sip.conf.
Later we start to monitor the call (writing *.wav files)
Dont know if that causes the described problems.
If the connection is made an the user on the other side of the line
takes the phone, we phone with x-pro.
Johann wrote:
Without information about your dialplan and what the phpagi script
does there is not much anyone can do. I do not know of any known
issues that may account for the problem you are having.
Update with further information and maybe someone will be able to
provide some insight.
--johann
Michael Häberle wrote:
Does nobody know a solution or an approach to a solution?
Michael
Michael Häberle wrote:
Hi there
In our php-application we use phpagi to communicate with asterisk
(as the voip-client we use x-pro)
Sometimes it occurs that the dialtone is very choppy or not present.
If we dial directly in x-pro this problem has never occured.
I dont know what the problem is, first I thought it is the bandwith
(which is actually a problem), but if that would be the major
problem it wouldnt work in x-pro either, I assume.
Another problem is that sometimes after two or three times ringing
the phone hangs up. No idea what the problem is. (this problem does
not occur with x-pro directly)
We use phpagi 2.14
Suse Linux 8.x
I dont know the asterix version (we downloaded it in july 2005)
Michael
_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Immosky AG
Service-Zentrale
Dufourstr. 5
CH-8702 Zollikon-Zürich
Tel +41 (0)43 344 52 52
Fax +41 (0)43 344 52 58
www.immosky.ch
[EMAIL PROTECTED]
_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
