zero massive
2007-Jan-02 13:17 UTC
[asterisk-users] Call connected, cannot hear or speak - $20 for fix
I am able to get this script to dial, but I am unable to talk or hear
anything. The script asks for the number to call and the the caller id to
display (if user is not at their normal extension). Once submitted, the
external extension receives a call, once answered the call is then placed to
the dentition number.
The script works as the call is place, but I cannot hear or say anything.
Any one that is able to get this going I would be will to give $20 to (via
paypal)
<html>
<head>
<title>call</title>
</head>
<body>
<?
#------------------------------------------------------------------------------------------
#edit the below variable values to reflect your system/information
#------------------------------------------------------------------------------------------
#specify the name/ip address of your asterisk box
#if your are hosting this page on your asterisk box, then you can use
#127.0.0.1 as the host IP. Otherwise, you will need to edit the following
#line in manager.conf, under the Admin user section:
#permit=127.0.0.1/255.255.255.0
#change to:
#permit=127.0.0.1/255.255.255.0,xxx.xxx.xxx.xxx ;(the ip address of the
server this page is running on)
$strHost = "127.0.0.1";
#specify the username you want to login with (these users are defined in
/etc/asterisk/manager.conf)
#this user is the default AAH AMP user; you shouldn't need to change, if
you're using AAH.
$strUser = "admin";
#specify the password for the above user
$strSecret = "amp111";
#specify the channel (extension) you want to receive the call requests with
#e.g. SIP/XXX, IAX2/XXXX, ZAP/XXXX, etc
$strChannel = "Local/15555555555@outrt-001-telasip";
#specify the context to make the outgoing call from. By default, AAH uses
from-internal
#Using from-internal will make you outgoing dialing rules apply
$strContext = "from-internal";
#specify the amount of time you want to try calling the specified channel
before hangin up
$strWaitTime = "30";
#specify the priority you wish to place on making this call
$strPriority = "1";
#specify the maximum amount of retries
$strMaxRetry = "2";
#--------------------------------------------------------------------------------------------
#Shouldn't need to edit anything below this point to make this script work
#--------------------------------------------------------------------------------------------
#get the phone number from the posted form
$strExten = $_POST['txtphonenumber'];
#specify the caller id for the call
$strCallerId = $_POST['txtcid'];
$length = strlen($strExten);
if ($length == 11 && is_numeric($strExten))
{
$oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die("Connection
to host failed");
fputs($oSocket, "Action: login\r\n");
fputs($oSocket, "Events: off\r\n");
fputs($oSocket, "Username: $strUser\r\n");
fputs($oSocket, "Secret: $strSecret\r\n\r\n");
fputs($oSocket, "Action: originate\r\n");
fputs($oSocket, "Channel: $strChannel\r\n");
fputs($oSocket, "WaitTime: $strWaitTime\r\n");
fputs($oSocket, "CallerId: $strCallerId\r\n");
fputs($oSocket, "Exten: $strExten\r\n");
fputs($oSocket, "Context: $strContext\r\n");
fputs($oSocket, "Priority: $strPriority\r\n\r\n");
fputs($oSocket, "Action: Logoff\r\n\r\n");
fclose($oSocket);
?>
<p>
<table width="300" border="1"
bordercolor="#630000" cellpadding="3"
cellspacing="0">
<tr><td>
<font size="2" face="verdana,georgia"
color="#630000">We are currently
trying to call you. Please be patient, and wait for
your phone to ring!<br>If your phone does not ring after 2 minutes, we
apologize, but must either be out, or
already on the
phone.<br><a href="<? echo $_SERVER['PHP_SELF']
?>">Try Again</a></font>
</td></tr>
</table>
</p>
<?
}
else
{
?>
<div align="center">
<table width="300" border="1"
bordercolor="#630000" cellpadding="3"
cellspacing="0">
<tr>
<td><form action="<? echo $_SERVER['PHP_SELF']
?>" method="post">
<p align="left"><font size="2"
face="verdana,arial,georgia"
color="#630000">Enter number to call (11
Digits):</font></p>
<p align="center">
<input type="text" size="20"
maxlength="11"
name="txtphonenumber">
</p>
<p><font size="2"
face="verdana,arial,georgia"
color="#630000">Enter your caller ID</font>:</p>
<p align="center">
<input type="text" size="20"
maxlength="11" name="txtcid">
</p>
<p>
<input type="submit" value="Make Call">
</p>
</form> </td></tr>
</table>
</p>
<?
}
?>
</div>
</body>
</html>
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Matt Riddell (NZ)
2007-Jan-07 14:16 UTC
[asterisk-users] Call connected, cannot hear or speak - $20 for fix
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 zero massive wrote:> I am able to get this script to dial, but I am unable to talk or hear > anything. The script asks for the number to call and the the caller id to > display (if user is not at their normal extension). Once submitted, the > external extension receives a call, once answered the call is then > placed to > the dentition number. > > The script works as the call is place, but I cannot hear or say anything. > Any one that is able to get this going I would be will to give $20 to (via > paypal)Does it work when you call from one of the phones to the other? Say in the script you are trying to connect User01 with User02. If you make a normal call between these users, is the audio passed? - -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFoWNCS6d5vy0jeVcRAoORAJ4zd5etQcQqntLSdxTWaCzqMwF78ACfW4Jz to2/ubpJXIU+7mQSVfvIIIM=e0/A -----END PGP SIGNATURE-----
Supa
2007-Jan-07 14:58 UTC
[asterisk-users] Call connected, cannot hear or speak - $20 for fix
Yes, it seems to fail when both extensions are external On 1/7/07, Matt Riddell (NZ) <matt.riddell@sineapps.com> wrote:> > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > zero massive wrote: > > I am able to get this script to dial, but I am unable to talk or hear > > anything. The script asks for the number to call and the the caller id > to > > display (if user is not at their normal extension). Once submitted, the > > external extension receives a call, once answered the call is then > > placed to > > the dentition number. > > > > The script works as the call is place, but I cannot hear or say > anything. > > Any one that is able to get this going I would be will to give $20 to > (via > > paypal) > > Does it work when you call from one of the phones to the other? > > Say in the script you are trying to connect User01 with User02. > > If you make a normal call between these users, is the audio passed? > > - -- > Cheers, > > Matt Riddell > _______________________________________________ > > http://www.sineapps.com/news.php (Daily Asterisk News - html) > http://wap.sineapps.com (Daily Asterisk News for your cellphone) > http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.2 (MingW32) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iD8DBQFFoWNCS6d5vy0jeVcRAoORAJ4zd5etQcQqntLSdxTWaCzqMwF78ACfW4Jz > to2/ubpJXIU+7mQSVfvIIIM> =e0/A > -----END PGP SIGNATURE----- > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070107/5abf83dd/attachment.htm
Matt Riddell (NZ)
2007-Jan-07 15:38 UTC
[asterisk-users] Call connected, cannot hear or speak - $20 for fix
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Supa wrote:> Yes, it seems to fail when both extensions are externalIt seems more like a NAT problem than a script problem. Are the phones both connected via SIP? - -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFoXZVS6d5vy0jeVcRAua7AJ9qOZuqlSEeqfWk5cG2v8nt7Buv5gCfVZSm to9odkqByRjDtwaKHogijEk=g4bZ -----END PGP SIGNATURE-----
Supa
2007-Jan-07 16:00 UTC
[asterisk-users] Call connected, cannot hear or speak - $20 for fix
I using my provdier like so SIP/Telasip-gw4/5198843344 when bridging calls. All my local extensions work, so does disa and the like On 1/7/07, Matt Riddell (NZ) <matt.riddell@sineapps.com> wrote:> > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1m usin > > Supa wrote: > > Yes, it seems to fail when both extensions are external > > It seems more like a NAT problem than a script problem. Are the phones > both connected via SIP? > > - -- > Cheers, > > Matt Riddell > _______________________________________________ > > http://www.sineapps.com/news.php (Daily Asterisk News - html) > http://wap.sineapps.com (Daily Asterisk News for your cellphone) > http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.2 (MingW32) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iD8DBQFFoXZVS6d5vy0jeVcRAua7AJ9qOZuqlSEeqfWk5cG2v8nt7Buv5gCfVZSm > to9odkqByRjDtwaKHogijEk> =g4bZ > -----END PGP SIGNATURE----- > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070107/bfc3bd2e/attachment.htm
Matt Riddell (NZ)
2007-Feb-28 21:38 UTC
[asterisk-users] Call connected, cannot hear or speak - $20 for fix
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Supa wrote:> I using my provdier like so SIP/Telasip-gw4/5198843344 when bridging calls. > All my local extensions work, so does disa and the likeDid you get this going? - -- Cheers, Matt Riddell Director _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF5ljXS6d5vy0jeVcRAtiuAJ9m5LOTjFDiPdm+Ux3Ic6nXAPRcaACcDHjC J5Gdt8Rc/BDfi33U8Bku85A=A2KZ -----END PGP SIGNATURE-----