Displaying 6 results from an estimated 6 matches for "hvardan71".
2010 May 11
1
asterisk-users Digest, Vol 70, Issue 23
Thanks Vardan,
I will like to know if this scenario can work when peer is not having fixed
ip and we use
host = nasir.server.com
?
also I have set insecure=invite,port
what if i use
insecure=no
thanks again.
Message: 24
Date: Tue, 11 May 2010 10:52:14 +0500
From: Vardan <hvardan71 at gmail.com>
Subject: Re: [asterisk-users] Dialing a SIP Peer without using
register strin
To: asterisk-users at lists.digium.com
Message-ID: <hsarab$ok7$1 at dough.gmane.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Remove username and secret and use IP authent...
2010 Dec 20
5
DIALSTATUS on CANCEL
Hello,
We have a strange situation (asterisk 1.6.2.14), where we get a result for
DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL.
This is the (relevant) test dialplan:
--------------------------------
[incoming-private]
exten => _X., n, Dial(SIP/1001,30)
exten => _X., n, NoOp(${DIALSTATUS})
exten => _X., n, Gosub(incoming-status,s-${DIALSTATUS},1)
[incoming-status]
exten
2010 May 12
2
asterisk-users Digest, Vol 70, Issue 25
...is
payasyougo.*problem is that i want the call to land in that context on
nasir.server.com, which works if i use register string. but without register
string call goes to default context on nasir.server.com
regards,
Nasir Javaid
Message: 19
Date: Tue, 11 May 2010 20:54:30 +0500
From: Vardan <hvardan71 at gmail.com>
Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 24
To: asterisk-users at lists.digium.com
Message-ID: <hsbujk$qk9$1 at dough.gmane.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello Nasir
I have some please.
Do so, it help.
Find all re...
2010 May 11
1
asterisk-users Digest, Vol 70, Issue 24
...is scenario can work when peer is not having fixed
> ip and we use
> host = nasir.server.com
> ?
> also I have set insecure=invite,port
>
> what if i use
> insecure=no
>
> thanks again.
>
> Message: 24
> Date: Tue, 11 May 2010 10:52:14 +0500
> From: Vardan <hvardan71 at gmail.com>
> Subject: Re: [asterisk-users] Dialing a SIP Peer without using
> register strin
> To: asterisk-users at lists.digium.com
> Message-ID: <hsarab$ok7$1 at dough.gmane.org>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Remove usern...
2010 Sep 16
5
a2billing
Hey there,
I am trying to setup a2billing on asterisk 1.6 , but ,when I try to access its web page I see the a2billing directories:Index of /a2billingNameLast modifiedSizeDescriptionParent Directory -admin/15-Sep-2010 19:19-agent/15-Sep-2010 19:21-common/15-Sep-2010 19:18-customer/15-Sep-2010 19:20-Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny8 with Suhosin-Patch Server at
Att,
Flavio Roberto
2010 May 18
3
About option U in Dial Ast version 1.6.2
Has any one used this?
U(x[^arg[^...]]):
x - Name of the subroutine to execute via Gosub
arg - Arguments for the Gosub routine
Execute via Gosub the routine <x> for the *called* channel before
connecting to the calling channel. Arguments can be specified to
the Gosub
using '^' as a delimiter. The Gosub routine can set the variable ${GO