anyone?
what should x-lite account be for guest user ?I tried guest but didnt work
________________________________________
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Tuesday, January 03, 2012 5:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call
for example if I am using x-lite as client, how to I connect as guest from
client ...I am allowing guests at asterisk server
________________________________________
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Tuesday, January 03, 2012 5:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call
thank you very much for this explanation, but my question does my client have
to be registered first, right? what do i Use to register ... there should be
information to register with using guest,
I got your idea about the security, and I can work with that ... but at cleints
I need to have information to log with?right?
________________________________________
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn
[da...@klaverstyn.com.au]
Sent: Tuesday, January 03, 2012 5:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call
Hi,
The point of SIP guest calls is that there is no username and password required
to make calls. If you have enabled guest calls then whatever extensions you
have allowed in the allocate default sip context anyone will be able to dial.
If you have in your sip.conf file
context=from-vsp ; Default context for incoming calls
allowguest=yes ; Allow or reject guest calls (default is yes)
and in your extensions.conf file
exten => 202,1,GotoIf($[${LEN(${CALLERID(name)})}=0]?2:3)
exten => 202,n,Set(CALLERID(NAME)=Guest SIP User)
exten => 202,n,Dial(SIP/202,30,r)
exten => 202,n,VoiceMail(202@default,us)
exten => 202,n,HangUp
... then anyone will be able to call 202.
The key is to make sure people cannot make trunk calls from the guest context.
-----Original Message-----
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Tuesday, 3 January 2012 9:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to make SIP guest call
Actaully I didnt find a good example how to configure the guest call in
asterisk other than allowGuest in SIP.conf, anybody have a good example for
that?
thanx
________________________________________
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Tuesday, January 03, 2012 5:08 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to make SIP guest call
Hi all,
If I am enabling the SIP Guest calls,
How can I make the call?
what my SIP clients information to make the call?
I mean what there username and password for guest call?
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
