search for: hvardan71

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