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