Greetings all, I've just stumbled upon your project and it may solve an issue we are having. I've just spent about 3 weeks getting to know asterisk just to discover I don't think it can do what I need. We have a project where we have incoming calls on a SIP channel. We need to do a direct forward of these calls to an outgoing channel based to a number which is from our database. Simple to do in asterisk, but the problem is that we cannot have these calls "connected" between the two lines. They have an automated message at the beginning that is being activated when we do the answer before the dial of the second number in asterisk. Out first idea is to bridge the incoming call directly to the outgoing call. The problem is that the incoming call cannot be "answered" and then we initiate the outgoing call. It needs to be a seamless bridge between the two calls. A nice feature would be to have a timer on the call. I saw a bounty for the timer feature, so I'm guessing (hoping) the bridging part can be done now. One other thought we are having is the ability to leave the incoming line "ringing" and dial the outgoing line until it is answered. At that time, answer the incoming and then bridge them together. So my question is: Can freeswitch do these things? Thanks and congratulations on the nice work!
I believe so, visit us on irc and we'll figure it out. On Fri, Apr 4, 2008 at 7:34 PM, Tim Meade <Tim.Meade@fusedware.com> wrote:
Greetings all,
I've just stumbled upon your project and it may solve an issue we are having.
I've just spent about 3 weeks getting to know asterisk just to discover I don't think it can do what I need.
We have a project where we have incoming calls on a SIP channel. We need to do a direct forward of these calls to an outgoing channel based to a number which is from our database. Simple to do in asterisk, but the problem is that we cannot have these calls "connected" between the two lines. They have an automated message at the beginning that is being activated when we do the answer before the dial of the second number in asterisk.
Out first idea is to bridge the incoming call directly to the outgoing call. The problem is that the incoming call cannot be "answered" and then we initiate the outgoing call. It needs to be a seamless bridge between the two calls. A nice feature would be to have a timer on the call. I saw a bounty for the timer feature, so I'm guessing (hoping) the bridging part can be done now.
One other thought we are having is the ability to leave the incoming line "ringing" and dial the outgoing line until it is answered. At that time, answer the incoming and then bridge them together.
So my question is: Can freeswitch do these things?
Thanks and congratulations on the nice work!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Tim, I can attest to the fact that FreeSWITCH is a great project with a really cool community. Right now the IRC channel is the best place to get quick, specific information on how to handle the kinds of things you're looking to do. One of the important things about FS is that the devs made extremely wise engineering decisions long before a single line of code was written. They made sure that FS would be extremely flexible in what it can do. One byproduct of that is that we get visitors asking, "Hey, can FreeSWITCH do this?" when no one here had thought of that before. In many cases it can indeed "do that," but it takes a little tweaking and some attention from the experts. Definitely hop on the IRC channel #freeswitch; the main devs generally are there during the day and some evenings. If you do get FS up and running, especially in a production environment, we would ask that you consider documenting your setup on the wiki. We hope to lower the barrier to entry for others with similar needs by having good documentation, but of course we also like to brag about what FS can do! :-) Thanks for checking out FreeSWITCH! -Michael, aka mercutioviz ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tim Meade Sent: Friday, April 04, 2008 5:34 PM To: 'freeswitch-users@lists.freeswitch.org' Subject: [Freeswitch-users] freeswitch as a SIP Bridge Greetings all, I've just stumbled upon your project and it may solve an issue we are having. I've just spent about 3 weeks getting to know asterisk just to discover I don't think it can do what I need. We have a project where we have incoming calls on a SIP channel. We need to do a direct forward of these calls to an outgoing channel based to a number which is from our database. Simple to do in asterisk, but the problem is that we cannot have these calls "connected" between the two lines. They have an automated message at the beginning that is being activated when we do the answer before the dial of the second number in asterisk. Out first idea is to bridge the incoming call directly to the outgoing call. The problem is that the incoming call cannot be "answered" and then we initiate the outgoing call. It needs to be a seamless bridge between the two calls. A nice feature would be to have a timer on the call. I saw a bounty for the timer feature, so I'm guessing (hoping) the bridging part can be done now. One other thought we are having is the ability to leave the incoming line "ringing" and dial the outgoing line until it is answered. At that time, answer the incoming and then bridge them together. So my question is: Can freeswitch do these things? Thanks and congratulations on the nice work!
having another look at your email now that I have more time to examine it. I think that is relatively simple for us to do unless I am misunderstanding. You want to make an inbound sip call that then results in an outbound call and you do not want to connect the audio until the moment that the far end party answers right? With freeswitch you have many choices when originating and bridging a call and it's possible to influence the behavior in many ways. On Sat, Apr 5, 2008 at 2:03 AM, Michael Collins <mcollins@fcnetwork.com> wrote:
Tim,
I can attest to the fact that FreeSWITCH is a great project with a really cool community. Right now the IRC channel is the best place to get quick, specific information on how to handle the kinds of things you're looking to do. One of the important things about FS is that the devs made extremely wise engineering decisions long before a single line of code was written. They made sure that FS would be extremely flexible in what it can do. One byproduct of that is that we get visitors asking, "Hey, can FreeSWITCH do this?" when no one here had thought of that before. In many cases it can indeed "do that," but it takes a little tweaking and some attention from the experts. Definitely hop on the IRC channel #freeswitch; the main devs generally are there during the day and some evenings.
If you do get FS up and running, especially in a production environment, we would ask that you consider documenting your setup on the wiki. We hope to lower the barrier to entry for others with similar needs by having good documentation, but of course we also like to brag about what FS can do! J
Thanks for checking out FreeSWITCH!
-Michael, aka mercutioviz
------------------------------
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Tim Meade *Sent:* Friday, April 04, 2008 5:34 PM *To:* 'freeswitch-users@lists.freeswitch.org' *Subject:* [Freeswitch-users] freeswitch as a SIP Bridge
Greetings all,
I've just stumbled upon your project and it may solve an issue we are having.
I've just spent about 3 weeks getting to know asterisk just to discover I don't think it can do what I need.
We have a project where we have incoming calls on a SIP channel. We need to do a direct forward of these calls to an outgoing channel based to a number which is from our database. Simple to do in asterisk, but the problem is that we cannot have these calls "connected" between the two lines. They have an automated message at the beginning that is being activated when we do the answer before the dial of the second number in asterisk.
Out first idea is to bridge the incoming call directly to the outgoing call. The problem is that the incoming call cannot be "answered" and then we initiate the outgoing call. It needs to be a seamless bridge between the two calls. A nice feature would be to have a timer on the call. I saw a bounty for the timer feature, so I'm guessing (hoping) the bridging part can be done now.
One other thought we are having is the ability to leave the incoming line "ringing" and dial the outgoing line until it is answered. At that time, answer the incoming and then bridge them together.
So my question is: Can freeswitch do these things?
Thanks and congratulations on the nice work!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Thanks Anthony. I'm going to have to give it a shot. Kind of looking forward to it. The issue with asterisk, just to keep everyone up to speed, is with my DID providers. We have three for testing: Broadvoice, VTWHITE, and DIDX. When the call comes in, we do NOT want it answered. The incoming call will always play an automated message and we do not want it to be played until a live human being answers the forwarded call. So the situation is: Incoming call should be left to ring. Outgoing call is initiated from our server to a known number. When outgoing call is answered, we bridge the two: OR we just bridge the incoming call to the outgoing while retaining control over the call. A couple of people here have mentioned that it should be easy to do. Thanks. Tim From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, April 05, 2008 11:46 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge having another look at your email now that I have more time to examine it. I think that is relatively simple for us to do unless I am misunderstanding. You want to make an inbound sip call that then results in an outbound call and you do not want to connect the audio until the moment that the far end party answers right? With freeswitch you have many choices when originating and bridging a call and it's possible to influence the behavior in many ways. On Sat, Apr 5, 2008 at 2:03 AM, Michael Collins <mcollins@fcnetwork.com<mailto:mcollins@fcnetwork.com>> wrote: Tim, I can attest to the fact that FreeSWITCH is a great project with a really cool community. Right now the IRC channel is the best place to get quick, specific information on how to handle the kinds of things you're looking to do. One of the important things about FS is that the devs made extremely wise engineering decisions long before a single line of code was written. They made sure that FS would be extremely flexible in what it can do. One byproduct of that is that we get visitors asking, "Hey, can FreeSWITCH do this?" when no one here had thought of that before. In many cases it can indeed "do that," but it takes a little tweaking and some attention from the experts. Definitely hop on the IRC channel #freeswitch; the main devs generally are there during the day and some evenings. If you do get FS up and running, especially in a production environment, we would ask that you consider documenting your setup on the wiki. We hope to lower the barrier to entry for others with similar needs by having good documentation, but of course we also like to brag about what FS can do! :) Thanks for checking out FreeSWITCH! -Michael, aka mercutioviz ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Tim Meade Sent: Friday, April 04, 2008 5:34 PM To: 'freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>' Subject: [Freeswitch-users] freeswitch as a SIP Bridge Greetings all, I've just stumbled upon your project and it may solve an issue we are having. I've just spent about 3 weeks getting to know asterisk just to discover I don't think it can do what I need. We have a project where we have incoming calls on a SIP channel. We need to do a direct forward of these calls to an outgoing channel based to a number which is from our database. Simple to do in asterisk, but the problem is that we cannot have these calls "connected" between the two lines. They have an automated message at the beginning that is being activated when we do the answer before the dial of the second number in asterisk. Out first idea is to bridge the incoming call directly to the outgoing call. The problem is that the incoming call cannot be "answered" and then we initiate the outgoing call. It needs to be a seamless bridge between the two calls. A nice feature would be to have a timer on the call. I saw a bounty for the timer feature, so I'm guessing (hoping) the bridging part can be done now. One other thought we are having is the ability to leave the incoming line "ringing" and dial the outgoing line until it is answered. At that time, answer the incoming and then bridge them together. So my question is: Can freeswitch do these things? Thanks and congratulations on the nice work! _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org<mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Thanks Michael. Well I'm going to give it a try. I'm off to find an IRC program now. I've identified the issues with asterisk and it's with my providers. Big question: I have a fedora and windows dev boxes. What's the best bet for someone new to the software? Thanks Tim From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, April 05, 2008 3:04 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge Tim, I can attest to the fact that FreeSWITCH is a great project with a really cool community. Right now the IRC channel is the best place to get quick, specific information on how to handle the kinds of things you're looking to do. One of the important things about FS is that the devs made extremely wise engineering decisions long before a single line of code was written. They made sure that FS would be extremely flexible in what it can do. One byproduct of that is that we get visitors asking, "Hey, can FreeSWITCH do this?" when no one here had thought of that before. In many cases it can indeed "do that," but it takes a little tweaking and some attention from the experts. Definitely hop on the IRC channel #freeswitch; the main devs generally are there during the day and some evenings. If you do get FS up and running, especially in a production environment, we would ask that you consider documenting your setup on the wiki. We hope to lower the barrier to entry for others with similar needs by having good documentation, but of course we also like to brag about what FS can do! :) Thanks for checking out FreeSWITCH! -Michael, aka mercutioviz ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tim Meade Sent: Friday, April 04, 2008 5:34 PM To: 'freeswitch-users@lists.freeswitch.org' Subject: [Freeswitch-users] freeswitch as a SIP Bridge Greetings all, I've just stumbled upon your project and it may solve an issue we are having. I've just spent about 3 weeks getting to know asterisk just to discover I don't think it can do what I need. We have a project where we have incoming calls on a SIP channel. We need to do a direct forward of these calls to an outgoing channel based to a number which is from our database. Simple to do in asterisk, but the problem is that we cannot have these calls "connected" between the two lines. They have an automated message at the beginning that is being activated when we do the answer before the dial of the second number in asterisk. Out first idea is to bridge the incoming call directly to the outgoing call. The problem is that the incoming call cannot be "answered" and then we initiate the outgoing call. It needs to be a seamless bridge between the two calls. A nice feature would be to have a timer on the call. I saw a bounty for the timer feature, so I'm guessing (hoping) the bridging part can be done now. One other thought we are having is the ability to leave the incoming line "ringing" and dial the outgoing line until it is answered. At that time, answer the incoming and then bridge them together. So my question is: Can freeswitch do these things? Thanks and congratulations on the nice work!
Tim, If you're comfortable with Linux then I would suggest starting there, but you are by no means locked in. Much of the development of FS is done in CentOS, and I believe in a 64-bit hardware environment. You'll also find that the current documentation is slanted toward a Linux/Unix environment. This is probably due to the fact that most FS users don't fall into the category of Microsoft fans. :-) However, if you're 100% VoIP then Windows should be fine. (Please tell me you're not on Vista!!) The PSTN/TDM stuff gets a bit tricky in Windows, so be on the lookout if you are thinking about future PSTN connectivity. Personally, I'm using CentOS 5.1 and I'm extremely happy with the performance and setup. For the record, if you choose Gentoo then you're quite likely to get yelled at! :-) Let us know how it goes. We'd be curious to see how FS stacks up in a head-to-head comparison against Asterisk in your application. Thanks again for checking it out, MC ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tim Meade Sent: Saturday, April 05, 2008 8:55 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge Thanks Michael. Well I'm going to give it a try. I'm off to find an IRC program now. I've identified the issues with asterisk and it's with my providers. Big question: I have a fedora and windows dev boxes. What's the best bet for someone new to the software? Thanks Tim From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, April 05, 2008 3:04 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge Tim, I can attest to the fact that FreeSWITCH is a great project with a really cool community. Right now the IRC channel is the best place to get quick, specific information on how to handle the kinds of things you're looking to do. One of the important things about FS is that the devs made extremely wise engineering decisions long before a single line of code was written. They made sure that FS would be extremely flexible in what it can do. One byproduct of that is that we get visitors asking, "Hey, can FreeSWITCH do this?" when no one here had thought of that before. In many cases it can indeed "do that," but it takes a little tweaking and some attention from the experts. Definitely hop on the IRC channel #freeswitch; the main devs generally are there during the day and some evenings. If you do get FS up and running, especially in a production environment, we would ask that you consider documenting your setup on the wiki. We hope to lower the barrier to entry for others with similar needs by having good documentation, but of course we also like to brag about what FS can do! :-) Thanks for checking out FreeSWITCH! -Michael, aka mercutioviz ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tim Meade Sent: Friday, April 04, 2008 5:34 PM To: 'freeswitch-users@lists.freeswitch.org' Subject: [Freeswitch-users] freeswitch as a SIP Bridge Greetings all, I've just stumbled upon your project and it may solve an issue we are having. I've just spent about 3 weeks getting to know asterisk just to discover I don't think it can do what I need. We have a project where we have incoming calls on a SIP channel. We need to do a direct forward of these calls to an outgoing channel based to a number which is from our database. Simple to do in asterisk, but the problem is that we cannot have these calls "connected" between the two lines. They have an automated message at the beginning that is being activated when we do the answer before the dial of the second number in asterisk. Out first idea is to bridge the incoming call directly to the outgoing call. The problem is that the incoming call cannot be "answered" and then we initiate the outgoing call. It needs to be a seamless bridge between the two calls. A nice feature would be to have a timer on the call. I saw a bounty for the timer feature, so I'm guessing (hoping) the bridging part can be done now. One other thought we are having is the ability to leave the incoming line "ringing" and dial the outgoing line until it is answered. At that time, answer the incoming and then bridge them together. So my question is: Can freeswitch do these things? Thanks and congratulations on the nice work!
The "How FreeSWITCH is Being Used" is still coming soon on the wiki. I'd be curious to see what configurations of OS/hardware people have and how they're using FreeSWITCH. Think it's worth adding a page on the wiki for this? - Dale ---------------------------------------------- http://myhelpa.com - The web when you need it. On Sat, 2008-04-05 at 21:37 -0700, Michael Collins wrote:
Tim,
If you’re comfortable with Linux then I would suggest starting there, but you are by no means locked in. Much of the development of FS is done in CentOS, and I believe in a 64-bit hardware environment. You’ll also find that the current documentation is slanted toward a Linux/Unix environment. This is probably due to the fact that most FS users don’t fall into the category of Microsoft fans. J However, if you’re 100% VoIP then Windows should be fine. (Please tell me you’re not on Vista!!) The PSTN/TDM stuff gets a bit tricky in Windows, so be on the lookout if you are thinking about future PSTN connectivity.
Personally, I’m using CentOS 5.1 and I’m extremely happy with the performance and setup. For the record, if you choose Gentoo then you’re quite likely to get yelled at! J
Let us know how it goes. We’d be curious to see how FS stacks up in a head-to-head comparison against Asterisk in your application.
Thanks again for checking it out,
MC
______________________________________________________________________ From:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tim Meade Sent: Saturday, April 05, 2008 8:55 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge
Thanks Michael.
Well I’m going to give it a try. I’m off to find an IRC program now.
I’ve identified the issues with asterisk and it’s with my providers.
Big question:
I have a fedora and windows dev boxes. What’s the best bet for someone new to the software?
Thanks
Tim
From:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, April 05, 2008 3:04 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge
Tim,
I can attest to the fact that FreeSWITCH is a great project with a really cool community. Right now the IRC channel is the best place to get quick, specific information on how to handle the kinds of things you’re looking to do. One of the important things about FS is that the devs made extremely wise engineering decisions long before a single line of code was written. They made sure that FS would be extremely flexible in what it can do. One byproduct of that is that we get visitors asking, “Hey, can FreeSWITCH do this?” when no one here had thought of that before. In many cases it can indeed “do that,” but it takes a little tweaking and some attention from the experts. Definitely hop on the IRC channel #freeswitch; the main devs generally are there during the day and some evenings.
If you do get FS up and running, especially in a production environment, we would ask that you consider documenting your setup on the wiki. We hope to lower the barrier to entry for others with similar needs by having good documentation, but of course we also like to brag about what FS can do! J
Thanks for checking out FreeSWITCH!
-Michael, aka mercutioviz
______________________________________________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tim Meade Sent: Friday, April 04, 2008 5:34 PM To: 'freeswitch-users@lists.freeswitch.org' Subject: [Freeswitch-users] freeswitch as a SIP Bridge
Greetings all,
I’ve just stumbled upon your project and it may solve an issue we are having.
I’ve just spent about 3 weeks getting to know asterisk just to discover I don’t think it can do what I need.
We have a project where we have incoming calls on a SIP channel. We need to do a direct forward of these calls to an outgoing channel based to a number which is from our database. Simple to do in asterisk, but the problem is that we cannot have these calls “connected” between the two lines. They have an automated message at the beginning that is being activated when we do the answer before the dial of the second number in asterisk.
Out first idea is to bridge the incoming call directly to the outgoing call. The problem is that the incoming call cannot be “answered” and then we initiate the outgoing call. It needs to be a seamless bridge between the two calls. A nice feature would be to have a timer on the call. I saw a bounty for the timer feature, so I’m guessing (hoping) the bridging part can be done now.
One other thought we are having is the ability to leave the incoming line “ringing” and dial the outgoing line until it is answered. At that time, answer the incoming and then bridge them together.
So my question is: Can freeswitch do these things?
Thanks and congratulations on the nice work!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Dale, When I added the link for "how fs is being used" I was envisioning that those posting would include their hardware and OS environments. In fact, I was kicking around having a small template for the info we'd all like to see included in all of the examples. E.g.: Number fs servers: Operating system: Hardware platform: (CPUs, RAM, RAID, NIC speed, etc.) TDM: Y/N; if yes, what kind of TDM card(s) Stuff like that. If you have any ideas on what else would make for good case studies or whatnot on "how fs is being used" then please let us know. Thanks! -MC
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Dale Thatcher Sent: Saturday, April 05, 2008 10:51 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] How FreeSWITCH is being used
The "How FreeSWITCH is Being Used" is still coming soon on the wiki. I'd be curious to see what configurations of OS/hardware people have and how they're using FreeSWITCH. Think it's worth adding a page on the wiki for this?
- Dale
---------------------------------------------- http://myhelpa.com - The web when you need it.
On Sat, 2008-04-05 at 21:37 -0700, Michael Collins wrote:
Tim,
If you're comfortable with Linux then I would suggest starting there, but you are by no means locked in. Much of the development of FS is done in CentOS, and I believe in a 64-bit hardware environment. You'll also find that the current documentation is slanted toward a Linux/Unix environment. This is probably due to the fact that most FS users don't fall into the category of Microsoft fans. J However, if you're 100% VoIP then Windows should be fine. (Please tell me you're not on Vista!!) The PSTN/TDM stuff gets a bit tricky in Windows, so be on the lookout if you are thinking about future PSTN connectivity.
Personally, I'm using CentOS 5.1 and I'm extremely happy with the performance and setup. For the record, if you choose Gentoo then you're quite likely to get yelled at! J
Let us know how it goes. We'd be curious to see how FS stacks up in a head-to-head comparison against Asterisk in your application.
Thanks again for checking it out,
MC
From:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tim Meade Sent: Saturday, April 05, 2008 8:55 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge
Thanks Michael.
Well I'm going to give it a try. I'm off to find an IRC program now.
I've identified the issues with asterisk and it's with my providers.
Big question:
I have a fedora and windows dev boxes. What's the best bet for someone new to the software?
Thanks
Tim
From:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, April 05, 2008 3:04 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge
Tim,
I can attest to the fact that FreeSWITCH is a great project with a really cool community. Right now the IRC channel is the best place to get quick, specific information on how to handle the kinds of things you're looking to do. One of the important things about FS is that the devs made extremely wise engineering decisions long before a single line of code was written. They made sure that FS would be extremely flexible in what it can do. One byproduct of that is that we get visitors asking, "Hey, can FreeSWITCH do this?" when no one here had thought of that before. In many cases it can indeed "do that," but it takes a little tweaking and some attention from the experts. Definitely hop on the IRC channel #freeswitch; the main devs generally are there during the day and some evenings.
If you do get FS up and running, especially in a production environment, we would ask that you consider documenting your setup on the wiki. We hope to lower the barrier to entry for others with similar needs by having good documentation, but of course we also like to brag about what FS can do! J
Thanks for checking out FreeSWITCH!
-Michael, aka mercutioviz
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tim Meade Sent: Friday, April 04, 2008 5:34 PM To: 'freeswitch-users@lists.freeswitch.org' Subject: [Freeswitch-users] freeswitch as a SIP Bridge
Greetings all,
I've just stumbled upon your project and it may solve an issue we are having.
I've just spent about 3 weeks getting to know asterisk just to discover I don't think it can do what I need.
We have a project where we have incoming calls on a SIP channel. We need to do a direct forward of these calls to an outgoing channel based to a number which is from our database. Simple to do in asterisk, but the problem is that we cannot have these calls "connected" between the two lines. They have an automated message at the beginning that is being activated when we do the answer before the dial of the second number in asterisk.
Out first idea is to bridge the incoming call directly to the outgoing call. The problem is that the incoming call cannot be "answered" and then we initiate the outgoing call. It needs to be a seamless bridge between the two calls. A nice feature would be to have a timer on the call. I saw a bounty for the timer feature, so I'm guessing (hoping) the bridging part can be done now.
One other thought we are having is the ability to leave the incoming line "ringing" and dial the outgoing line until it is answered. At that time, answer the incoming and then bridge them together.
So my question is: Can freeswitch do these things?
Thanks and congratulations on the nice work!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Looks like a good starting point, how about adding a brief description of the application(s)? One of the great things about FreeSWITCH is it's flexibility so I'm sure a lot of people are using it in very different ways. Why not stick your template up and then people can start adding their info? thanks, - Dale On Sun, 2008-04-06 at 23:07 -0700, Michael Collins wrote:
Dale,
When I added the link for "how fs is being used" I was envisioning that those posting would include their hardware and OS environments. In fact, I was kicking around having a small template for the info we'd all like to see included in all of the examples.
E.g.: Number fs servers: Operating system: Hardware platform: (CPUs, RAM, RAID, NIC speed, etc.) TDM: Y/N; if yes, what kind of TDM card(s)
Stuff like that. If you have any ideas on what else would make for good case studies or whatnot on "how fs is being used" then please let us know.
Thanks!
-MC
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Dale Thatcher Sent: Saturday, April 05, 2008 10:51 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] How FreeSWITCH is being used
The "How FreeSWITCH is Being Used" is still coming soon on the wiki. I'd be curious to see what configurations of OS/hardware people have and how they're using FreeSWITCH. Think it's worth adding a page on the wiki for this?
- Dale
---------------------------------------------- http://myhelpa.com - The web when you need it.
On Sat, 2008-04-05 at 21:37 -0700, Michael Collins wrote:
Tim,
If you're comfortable with Linux then I would suggest starting there, but you are by no means locked in. Much of the development of FS is done in CentOS, and I believe in a 64-bit hardware environment. You'll also find that the current documentation is slanted toward a Linux/Unix environment. This is probably due to the fact that most FS users don't fall into the category of Microsoft fans. J However, if you're 100% VoIP then Windows should be fine. (Please tell me you're not on Vista!!) The PSTN/TDM stuff gets a bit tricky in Windows, so be on the lookout if you are thinking about future PSTN connectivity.
Personally, I'm using CentOS 5.1 and I'm extremely happy with the performance and setup. For the record, if you choose Gentoo then you're quite likely to get yelled at! J
Let us know how it goes. We'd be curious to see how FS stacks up in a head-to-head comparison against Asterisk in your application.
Thanks again for checking it out,
MC
From:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tim Meade Sent: Saturday, April 05, 2008 8:55 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge
Thanks Michael.
Well I'm going to give it a try. I'm off to find an IRC program now.
I've identified the issues with asterisk and it's with my providers.
Big question:
I have a fedora and windows dev boxes. What's the best bet for someone new to the software?
Thanks
Tim
From:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, April 05, 2008 3:04 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge
Tim,
I can attest to the fact that FreeSWITCH is a great project with a really cool community. Right now the IRC channel is the best place to get quick, specific information on how to handle the kinds of things you're looking to do. One of the important things about FS is that the devs made extremely wise engineering decisions long before a single line of code was written. They made sure that FS would be extremely flexible in what it can do. One byproduct of that is that we get visitors asking, "Hey, can FreeSWITCH do this?" when no one here had thought of that before. In many cases it can indeed "do that," but it takes a little tweaking and some attention from the experts. Definitely hop on the IRC channel #freeswitch; the main devs generally are there during the day and some evenings.
If you do get FS up and running, especially in a production environment, we would ask that you consider documenting your setup on the wiki. We hope to lower the barrier to entry for others with similar needs by having good documentation, but of course we also like to brag about what FS can do! J
Thanks for checking out FreeSWITCH!
-Michael, aka mercutioviz
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tim Meade Sent: Friday, April 04, 2008 5:34 PM To: 'freeswitch-users@lists.freeswitch.org' Subject: [Freeswitch-users] freeswitch as a SIP Bridge
Greetings all,
I've just stumbled upon your project and it may solve an issue we are having.
I've just spent about 3 weeks getting to know asterisk just to discover I don't think it can do what I need.
We have a project where we have incoming calls on a SIP channel. We need to do a direct forward of these calls to an outgoing channel based to a number which is from our database. Simple to do in asterisk, but the problem is that we cannot have these calls "connected" between the two lines. They have an automated message at the beginning that is being activated when we do the answer before the dial of the second number in asterisk.
Out first idea is to bridge the incoming call directly to the outgoing call. The problem is that the incoming call cannot be "answered" and then we initiate the outgoing call. It needs to be a seamless bridge between the two calls. A nice feature would be to have a timer on the call. I saw a bounty for the timer feature, so I'm guessing (hoping) the bridging part can be done now.
One other thought we are having is the ability to leave the incoming line "ringing" and dial the outgoing line until it is answered. At that time, answer the incoming and then bridge them together.
So my question is: Can freeswitch do these things?
Thanks and congratulations on the nice work!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
http://myhelpa.com - The web when you need it.
Well, the short answer is, "Cuz I don't know how!" :P I'll get with Brian and see what we can come up with. Thanks for the suggestions, MC
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Dale Thatcher Sent: Monday, April 07, 2008 12:20 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] How FreeSWITCH is being used
Looks like a good starting point, how about adding a brief description of the application(s)? One of the great things about FreeSWITCH is it's flexibility so I'm sure a lot of people are using it in very different ways.
Why not stick your template up and then people can start adding their info?
thanks,
- Dale
On Sun, 2008-04-06 at 23:07 -0700, Michael Collins wrote:
Dale,
When I added the link for "how fs is being used" I was envisioning that those posting would include their hardware and OS environments. In fact, I was kicking around having a small template for the info we'd all like to see included in all of the examples.
E.g.: Number fs servers: Operating system: Hardware platform: (CPUs, RAM, RAID, NIC speed, etc.) TDM: Y/N; if yes, what kind of TDM card(s)
Stuff like that. If you have any ideas on what else would make for good case studies or whatnot on "how fs is being used" then please let us know.
Thanks!
-MC
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Dale Thatcher Sent: Saturday, April 05, 2008 10:51 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] How FreeSWITCH is being used
The "How FreeSWITCH is Being Used" is still coming soon on the wiki. I'd be curious to see what configurations of OS/hardware people have and how they're using FreeSWITCH. Think it's worth adding a page on the wiki for this?
- Dale
---------------------------------------------- http://myhelpa.com - The web when you need it.
On Sat, 2008-04-05 at 21:37 -0700, Michael Collins wrote:
Tim,
If you're comfortable with Linux then I would suggest starting there, but you are by no means locked in. Much of the development of FS is done in CentOS, and I believe in a 64-bit hardware environment. You'll also find that the current documentation is slanted toward a Linux/Unix environment. This is probably due to the fact that most FS users don't fall into the category of Microsoft fans. J However, if you're 100% VoIP then Windows should be fine. (Please tell me you're not on Vista!!) The PSTN/TDM stuff gets a bit tricky in Windows, so be on the lookout if you are thinking about future PSTN connectivity.
Personally, I'm using CentOS 5.1 and I'm extremely happy with the performance and setup. For the record, if you choose Gentoo then you're quite likely to get yelled at! J
Let us know how it goes. We'd be curious to see how FS stacks up in a head-to-head comparison against Asterisk in your application.
Thanks again for checking it out,
MC
From:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tim Meade Sent: Saturday, April 05, 2008 8:55 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge
Thanks Michael.
Well I'm going to give it a try. I'm off to find an IRC program now.
I've identified the issues with asterisk and it's with my providers.
Big question:
I have a fedora and windows dev boxes. What's the best bet for someone new to the software?
Thanks
Tim
From:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, April 05, 2008 3:04 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge
Tim,
I can attest to the fact that FreeSWITCH is a great project with a really cool community. Right now the IRC channel is the best place to get quick, specific information on how to handle the kinds of things you're looking to do. One of the important things about FS is that the devs made extremely wise engineering decisions long before a single line of code was written. They made sure that FS would be extremely flexible in what it can do. One byproduct of that is that we get visitors asking, "Hey, can FreeSWITCH do this?" when no one here had thought of that before. In many cases it can indeed "do that," but it takes a little tweaking and some attention from the experts. Definitely hop on the IRC channel #freeswitch; the main devs generally are there during the day and some evenings.
If you do get FS up and running, especially in a production environment, we would ask that you consider documenting your setup on the wiki. We hope to lower the barrier to entry for others with similar needs by having good documentation, but of course we also like to brag about what FS can do! J
Thanks for checking out FreeSWITCH!
-Michael, aka mercutioviz
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Tim Meade Sent: Friday, April 04, 2008 5:34 PM To: 'freeswitch-users@lists.freeswitch.org' Subject: [Freeswitch-users] freeswitch as a SIP Bridge
Greetings all,
I've just stumbled upon your project and it may solve an issue we are having.
I've just spent about 3 weeks getting to know asterisk just to discover I don't think it can do what I need.
We have a project where we have incoming calls on a SIP channel. We need to do a direct forward of these calls to an outgoing channel based to a number which is from our database. Simple to do in asterisk, but the problem is that we cannot have these calls "connected" between the two lines. They have an automated message at the beginning that is being activated when we do the answer before the dial of the second number in asterisk.
Out first idea is to bridge the incoming call directly to the outgoing call. The problem is that the incoming call cannot be "answered" and then we initiate the outgoing call. It needs to be a seamless bridge between the two calls. A nice feature would be to have a timer on the call. I saw a bounty for the timer feature, so I'm guessing (hoping) the bridging part can be done now.
One other thought we are having is the ability to leave the incoming line "ringing" and dial the outgoing line until it is answered. At that time, answer the incoming and then bridge them together.
So my question is: Can freeswitch do these things?
Thanks and congratulations on the nice work!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://myhelpa.com - The web when you need it.
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Hi, I am looking for a little guidance. We have built our system around Asterisk and now we are looking to FreeSwitch to see if it will do what we need. I am confident that it can, but I need a tiny bit of pointing in the right direction. That said, here is what we do now: We do what we call Three Party Calling (call leg A, announce their minute balance, call leg B, bridge) Pretty easy to do really. The part I need help with is how we can get the balance to be announced as that data is contained in our SQL server database. Ideally, this data would come from the database but by way of C#. We also would like to get FreeSwitch to tell us the CDR data, http is fine if need be. I know #freeswitch is available, but I thought I would ask here first. Thanks! Christian
I would use mod_xml_curl to query your C# server after a dialplan for each call. Then you can create dialplan actions that answers, says the balance and then bridge to the destination. Or you can create a small javascript that answers, uses curl to fetch balance from a http server (or use the socket module, or odbc to query the db directly) and then bridge the call to the destination. As for CDR I would recommend the mod_cdr_xml module which sends CDR XML to a http server. I'm also coding a ODBC CDR module, but I havent had time to finish it :( On Sun, Apr 6, 2008 at 9:02 AM, Christian Jensen <christian@verbx.com> wrote:
Hi,
I am looking for a little guidance. We have built our system around Asterisk and now we are looking to FreeSwitch to see if it will do what we need. I am confident that it can, but I need a tiny bit of pointing in the right direction.
That said, here is what we do now:
We do what we call Three Party Calling (call leg A, announce their minute balance, call leg B, bridge)
Pretty easy to do really.
The part I need help with is how we can get the balance to be announced as that data is contained in our SQL server database. Ideally, this data would come from the database but by way of C#. We also would like to get FreeSwitch to tell us the CDR data, http is fine if need be.
I know #freeswitch is available, but I thought I would ask here first.
Thanks! Christian
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Also we are almost ready to revamp the mod_mono we have in tree that wold allow you to embed c# apps like mod_java and mod_python do. On Sun, Apr 6, 2008 at 12:28 PM, Jonas Gauffin <jonas.gauffin@gmail.com> wrote:
I would use mod_xml_curl to query your C# server after a dialplan for each call. Then you can create dialplan actions that answers, says the balance and then bridge to the destination. Or you can create a small javascript that answers, uses curl to fetch balance from a http server (or use the socket module, or odbc to query the db directly) and then bridge the call to the destination.
As for CDR I would recommend the mod_cdr_xml module which sends CDR XML to a http server. I'm also coding a ODBC CDR module, but I havent had time to finish it :(
On Sun, Apr 6, 2008 at 9:02 AM, Christian Jensen <christian@verbx.com> wrote:
Hi,
I am looking for a little guidance. We have built our system around Asterisk and now we are looking to FreeSwitch to see if it will do what we need. I am confident that it can, but I need a tiny bit of pointing in the right direction.
That said, here is what we do now:
We do what we call Three Party Calling (call leg A, announce their minute balance, call leg B, bridge)
Pretty easy to do really.
The part I need help with is how we can get the balance to be announced as that data is contained in our SQL server database. Ideally, this data would come from the database but by way of C#. We also would like to get FreeSwitch to tell us the CDR data, http is fine if need be.
I know #freeswitch is available, but I thought I would ask here first.
Thanks! Christian
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Those are great suggestions! Thanks! Any idea on how best to initiate leg A? Christian On Apr 6, 2008, at 10:28 AM, Jonas Gauffin wrote:
I would use mod_xml_curl to query your C# server after a dialplan for each call. Then you can create dialplan actions that answers, says the balance and then bridge to the destination. Or you can create a small javascript that answers, uses curl to fetch balance from a http server (or use the socket module, or odbc to query the db directly) and then bridge the call to the destination.
As for CDR I would recommend the mod_cdr_xml module which sends CDR XML to a http server. I'm also coding a ODBC CDR module, but I havent had time to finish it :(
On Sun, Apr 6, 2008 at 9:02 AM, Christian Jensen <christian@verbx.com> wrote:
Hi,
I am looking for a little guidance. We have built our system around Asterisk and now we are looking to FreeSwitch to see if it will do what we need. I am confident that it can, but I need a tiny bit of pointing in the right direction.
That said, here is what we do now:
We do what we call Three Party Calling (call leg A, announce their minute balance, call leg B, bridge)
Pretty easy to do really.
The part I need help with is how we can get the balance to be announced as that data is contained in our SQL server database. Ideally, this data would come from the database but by way of C#. We also would like to get FreeSwitch to tell us the CDR data, http is fine if need be.
I know #freeswitch is available, but I thought I would ask here first.
Thanks! Christian
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Christian, I'd do it differently: I'm afraid that's one of the things about FS, that there's often no single right way of doing something. CDRs - I'd use mod_xml_cdr to write them to files, and then have something pull the files from the directory and deal with them. You're a bit more fault-tolerant that way than with Jonas' suggestion. Let me know if you'd like a bit of Perl which pulls them from the directory and stashes them in a database - modifying it to post them to a web server is trivial. And the script itself - I'd write it in Perl and attach it to FreeSWITCH's event socket interface, possibly using a socket-based (HTTP?) interface to query your SQL server via a bit of C#, given the constraints that you've specified. DO NOT use FreeSWITCH::Client.pm as a basis for the script; it's crashed periodically in not terribly heavy use. Again, let me know if you want a framework that does all the FS-specific bits: it's pretty alpha, but we're using it in production. Hope this helps more than it confuses ;-) Cheers -- Dave
Those are great suggestions! Thanks!
Any idea on how best to initiate leg A?
Christian
On Apr 6, 2008, at 10:28 AM, Jonas Gauffin wrote:
I would use mod_xml_curl to query your C# server after a dialplan for each call. Then you can create dialplan actions that answers, says the balance and then bridge to the destination. Or you can create a small javascript that answers, uses curl to fetch balance from a http server (or use the socket module, or odbc to query the db directly) and then bridge the call to the destination.
As for CDR I would recommend the mod_cdr_xml module which sends CDR XML to a http server. I'm also coding a ODBC CDR module, but I havent had time to finish it :(
On Sun, Apr 6, 2008 at 9:02 AM, Christian Jensen <christian@verbx.com> wrote:
Hi,
I am looking for a little guidance. We have built our system around Asterisk and now we are looking to FreeSwitch to see if it will do what we need. I am confident that it can, but I need a tiny bit of pointing in the right direction.
That said, here is what we do now:
We do what we call Three Party Calling (call leg A, announce their minute balance, call leg B, bridge)
Pretty easy to do really.
The part I need help with is how we can get the balance to be announced as that data is contained in our SQL server database. Ideally, this data would come from the database but by way of C#. We also would like to get FreeSwitch to tell us the CDR data, http is fine if need be.
I know #freeswitch is available, but I thought I would ask here first.
Thanks! Christian
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk
participants (7)
-
Anthony Minessale
-
Christian Jensen
-
Dale Thatcher
-
David Knell
-
Jonas Gauffin
-
Michael Collins
-
Tim Meade