[asterisk-users] Connecting Skype to Asterisk
Hi all, I have an Asterisk PBX under development, that I would like to link to a Skype account if possible. The idea is that people would call a particular Skype username, and be redirected to my SIP and through that to Asterisk. Is this doable? I have looked around and saw the Skype for Asterisk driver, but of course that has been discontinued. Are there any other options? I would prefer not to have to go through the regular PSTN telephone network but directly from Skype to Asterisk via SIP. If you have any tips on how to configure my sip.conf to get this working, this would also be highly appreciated. Thanks in advance for your help! Kind regards, Philip Bennefall -- _ -- 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
Re: [asterisk-users] Connecting Skype to Asterisk
From what I gather, it costs extra for each channel even for direct Skype to Asterisk calls. Since my plan was to use this for business purposes, I'd need at least something like 30 channels which would be way out of my monthly budget unfortunately. Kind regards, Philip Bennefall - Original Message - From: Duncan Turnbull dun...@e-simple.co.nz To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: phi...@blastbay.com Sent: Friday, October 12, 2012 11:08 PM Subject: Re: [asterisk-users] Connecting Skype to Asterisk On 13/10/2012, at 7:54 AM, Christopher Harrington ch...@acsdi.com wrote: On Fri, Oct 12, 2012 at 1:44 PM, Philip Bennefall phi...@blastbay.com wrote: Hi all, I have an Asterisk PBX under development, that I would like to link to a Skype account if possible. The idea is that people would call a particular Skype username, and be redirected to my SIP and through that to Asterisk. Is this doable? I have looked around and saw the Skype for Asterisk driver, but of course that has been discontinued. Are there any other options? I would prefer not to have to go through the regular PSTN telephone network but directly from Skype to Asterisk via SIP. If you have any tips on how to configure my sip.conf to get this working, this would also be highly appreciated. It looks like this is what you want: http://www.skype.com/intl/en/business/skype-connect/ This is pretty straight forward to use for inbound skype business user names and outbound either to pstn, skype numbers are a little more to setup There is a monthly cost but its not much and if you have skype users out there its a good way for them to connect in -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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
[asterisk-users] Regarding caller ID and security
Hi all, I am new to Asterisk, and would like to begin by saying that it is an absolutely fantastic system. Seems incredibly stable, well tested, and easy to use. Now, to my question. I am making a mix between a personal ads and a voicemail service, where I want each user to be able to submit an ad that others can respond to by recording messages that go into this users inbox. My original thought was to base this purely on the CALLERID(num) value, but quickly discovered that this is a bit unreliable. Sometimes when I would call in it'd say anonymous, other times it would give me a bunch of zero's, other times it would show me my real phone number, and once it actually gave me just random digits. I do have a wait call after answering but before my first soundf ile is triggered, in my pickup context. I am wondering what the best way to approach this is? Do I ask the user to enter their phone number, and then generate a code based upon this that will then serve as a password when you call back? Do I attempt to use CALLERID(num) to detect returning users, or is this not adviseable from a security perspective? Preferably, I would like to avoid using a code altogether but I am told that it is relatively easy to spoof phone numbers to hack into someone else's inbox. Note that I do not plan to allow direct SIP calls, only through a PSTN/SIP provider where the IP address is on a whitelist. Any tips on how to approach this would be highly appreciated. Basically I want to make it as easy as possible for my users, but maintain high security. Thanks in advance for any help, and thanks once again to the developers of Asterisk for making such an excellent tool! Kind regards, Philip Bennefall P.S. I also wanted to know whether there is a function to check if a string contains only digits? This would be useful as a sanity check before I look up the phone number in the MySql database, if I do decide to use CALLERID(num) in this way.-- _ -- 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
