Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
It is that type of mechanism that enum uses and yes it was to solve a similar goal, but in this case you need a 'route server' type system - in particular as this is for IP routing of PSTN end points not on an IP network. A discussion about this came up a while ago. I suggested something along the lines of BGP, where each endpoint announces prefixes of what they can get to. You'll need a central machine that everyone peers up with and then you can use a switch = statement or exten = _.,1,Dial in * to query that machine and get the best route for your call. If you make sure that your destination machines are not behind NAT or a firewall, you can do an IAX handoff to get the connection set peer to peer instead of through the central server. Example: 4 remote * machines, each configured with our BGP software. Machine 1 announces that it can terminate calls to country code 1 with a cost of .02. Machine 2 announces that it can terminate calls to 1 with a cost of .05. Machine 3 announces that it can terminate calls to 1-830 with a cost of 0. Machine 4 announces that it can terminate calls to 1-830-751 with a cost of 0. You place a call to 1-830-751-2000 and the system determines that it can place that call for a cost of 0 to machine 4. You place a call to 1-240-988-4000 and the system determines that it can place that call via either machine 1 or 2, but lowest cost is machine 1. [general summary to all branches of this thread] Yes, that describes TRIP (RFC3219 - http://www.zvon.org/tmRFC/RFC3219/Output/index.html) fairly accurately. While not having quite a central machine with which everyone peers, it may be that each ITAD (Internet Telephony Administrative Domain, like an ASN) would have one main router to which all their local Asterisk servers would be leafs, and then that core router would peer with other core routers at other ITADs or maybe some large IRR-like servers which were clearinghouse-only style route distributors. I offered money here on this list previously to anyone who thinks that they're qualified to develop and integrate a TRIP implementation into Asterisk. Warning: it's not a trivial issue, and I will only consider programmers with a full understanding of the magnitude of the task. This could threaten to be a surprisingly large mesh with possibly hundreds of thousands or millions of routes of an extremely dynamic nature, and such an implementation is not for the beginner. I'm still taking applications. In other notes: I saw in other parts of this thread the discussion about how to do number routing via DNS. This is a good idea, so good, in fact, that it already exists in Asterisk and is a set of RFCs. It's called ENUM, and it routes phone numbers via the DNS. See enum.conf and show application EnumLookup - the good folks at nic.at were kind enough to pay for and work on these improvements to Asterisk. ENUM is great, but it's going slowly as far as the hyper adoption rates of Internet time are any comparison. The main issues seem to be political, since the triple whammy of ownership, authorization, and administration seem to be in the way. If you are in a country that hates VoIP, don't expect to see above-board ENUM any time in the near future. :-( BUT: The nice thing about ENUM, especially in Asterisk (and hopefully soon in SER) is that one can specify cascading trees in which to look up data that are not necessarily e164.arpa. as the root. I will leave it to the reader to figure out why this is a good thing and a bad thing at the same time. ENUM and TRIP provide DIFFERENT functions: ENUM gives out exact answers, and TRIP provides gateway answers. First, you look up the number in ENUM. Is there an answer? If so, send call to that SIP/H323/IAX gateway. If no answer, then look up the number in TRIP and find someone who has a cheap/good/fast/whatever gateway to that particular number range, and send the call to that SIP/H323/IAX/etc. gateway. In fact, I had a really nasty thought the other day: make a DNS resolver hack that allows ENUM lookups to incorporate TRIP replies. Yuck, yuck, yuck... but it would allow TRIP integration into any system that supports ENUM with no additional work on the telephony client side. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
On Sat, 2003-10-04 at 18:53, Rich Adamson wrote: Why not add an Article to the www.voip-info.org site, and those that are interested with helping can list their FWD, IAXTEL, or other access number, probable hours of availability, any special focus skills, size of their current * environment, etc? I'm game. Sounds good. Wouldn't it be possible to login to the queue of an * server providing this servers from my extension through my * installation? That way calls could be routed to available volunteers. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Graadt van Roggenweg 328 [EMAIL PROTECTED] 3531 AH Utrecht tel: +31 (0)30 298 2255Postbus 19127 fax: +31 (0)30 298 21113501 DC Utrecht signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
On Sat, 4 Oct 2003, James Sharp wrote: Actually, if this was to be done, it might be an idea to do it with DNS, so client machines would just do Dial(IAX2/[EMAIL PROTECTED]/442071234567) and the DNS system would resolve which machine is the correct target - no cleverness at all required at the client end, so implementation would be portable across all the other gnophones etc. Yup. That would be the way to do it. I'll contribute the DNS code for it. Isn't that what e.164 was invented for? -- Jon Stockill [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
On Sat, 4 Oct 2003, James Sharp wrote: Actually, if this was to be done, it might be an idea to do it with DNS, so client machines would just do Dial(IAX2/[EMAIL PROTECTED]/442071234567) and the DNS system would resolve which machine is the correct target - no cleverness at all required at the client end, so implementation would be portable across all the other gnophones etc. Yup. That would be the way to do it. I'll contribute the DNS code for it. Isn't that what e.164 was invented for? It is that type of mechanism that enum uses and yes it was to solve a similar goal, but in this case you need a 'route server' type system - in particular as this is for IP routing of PSTN end points not on an IP network. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
It is that type of mechanism that enum uses and yes it was to solve a similar goal, but in this case you need a 'route server' type system - in particular as this is for IP routing of PSTN end points not on an IP network. A discussion about this came up a while ago. I suggested something along the lines of BGP, where each endpoint announces prefixes of what they can get to. You'll need a central machine that everyone peers up with and then you can use a switch = statement or exten = _.,1,Dial in * to query that machine and get the best route for your call. If you make sure that your destination machines are not behind NAT or a firewall, you can do an IAX handoff to get the connection set peer to peer instead of through the central server. Example: 4 remote * machines, each configured with our BGP software. Machine 1 announces that it can terminate calls to country code 1 with a cost of .02. Machine 2 announces that it can terminate calls to 1 with a cost of .05. Machine 3 announces that it can terminate calls to 1-830 with a cost of 0. Machine 4 announces that it can terminate calls to 1-830-751 with a cost of 0. You place a call to 1-830-751-2000 and the system determines that it can place that call for a cost of 0 to machine 4. You place a call to 1-240-988-4000 and the system determines that it can place that call via either machine 1 or 2, but lowest cost is machine 1. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Let's TALK ABOUT IT!!!
Everyone seems to be working on their own servers that are in there homes, offices and elsewhere. The only common thread is this list-serv. I haveseveral * servers set up here in Austin, Texas. I propose we set up one of them with with all the list-serv members. At this time the calls would be limited to on-net calls only to other members. As the project grows and other members add servers we will decide if we want to give all servers PSTN access. This will give everyone a realtime enviroment in which to test their ideas, as well as being able to talk directly to other members if walk through TECH SUPPORT is needed. ALSO, we can show Martin and Mark that tech support doesn't have to be in AL. So "Let's TALK ABOUT IT!!!" This message was checked by MailScan for WorkgroupMail. www.workgroupmail.com
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
sip wrote: Everyone seems to be working on their own servers that are in there homes, offices and elsewhere. The only common thread is this list-serv. I have several * servers set up here in Austin, Texas. I propose we set up one of them with with all the list-serv members. At this time the calls would be limited to on-net calls only to other members. Isn't this exactly what IAXTEL was supposed to be all about? or are you proposing somthing else? As the project grows and other members add servers we will decide if we want to give all servers PSTN access. Billing would be a major issue wih this, but its a cool idea.. This will give everyone a realtime enviroment in which to test their ideas, as well as being able to talk directly to other members if walk through TECH SUPPORT is needed. Sounds good.. ALSO, we can show Martin and Mark that tech support doesn't have to be in AL. I have to say the Digium tech support is pretty good.. I mailed in a query at 16:30 yesterday, AL time, and got a responce within about an hour.. So Let's TALK ABOUT IT!!! Yes.. lets.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
IAXTEL is a 1-700 system designed for on-net calls. We have very low-cost PSTN lines to 48 states for no-cost long-distance dialing. I plan to add some of these lines to the system. As servers are added in other cities around the world I envision dialing Berlin, Germany from Austin, Texas and using a local hop-off in Berlin to make a local call FREE. (No 2 cents per minute) If we could blend IAXTEL into our PSTN service that would be one major hurtle overcome. But that would put a programing burden on the guys at Digium and they seem to have their plate full. So Let's TALK ABOUT IT!!! - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 04, 2003 8:10 AM Subject: Re: [Asterisk-Users] Let's TALK ABOUT IT!!! sip wrote: Everyone seems to be working on their own servers that are in there homes, offices and elsewhere. The only common thread is this list-serv. I have several * servers set up here in Austin, Texas. I propose we set up one of them with with all the list-serv members. At this time the calls would be limited to on-net calls only to other members. Isn't this exactly what IAXTEL was supposed to be all about? or are you proposing somthing else? As the project grows and other members add servers we will decide if we want to give all servers PSTN access. Billing would be a major issue wih this, but its a cool idea.. This will give everyone a realtime enviroment in which to test their ideas, as well as being able to talk directly to other members if walk through TECH SUPPORT is needed. Sounds good.. ALSO, we can show Martin and Mark that tech support doesn't have to be in AL. I have to say the Digium tech support is pretty good.. I mailed in a query at 16:30 yesterday, AL time, and got a responce within about an hour.. So Let's TALK ABOUT IT!!! Yes.. lets.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This message was checked by MailScan for WorkgroupMail. www.workgroupmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
I welocme the idea.After all we are deploying and using a voice based technology so we can have something like Voice Mailing List where we will be having voice messages and realtime talks with other programmers and developers. I think this should set a precedent for other mailing list also. Rgds Manoj K Gupta - Original Message - From: sip [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 04, 2003 7:31 PM Subject: Re: [Asterisk-Users] Let's TALK ABOUT IT!!! IAXTEL is a 1-700 system designed for on-net calls. We have very low-cost PSTN lines to 48 states for no-cost long-distance dialing. I plan to add some of these lines to the system. As servers are added in other cities around the world I envision dialing Berlin, Germany from Austin, Texas and using a local hop-off in Berlin to make a local call FREE. (No 2 cents per minute) If we could blend IAXTEL into our PSTN service that would be one major hurtle overcome. But that would put a programing burden on the guys at Digium and they seem to have their plate full. So Let's TALK ABOUT IT!!! - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 04, 2003 8:10 AM Subject: Re: [Asterisk-Users] Let's TALK ABOUT IT!!! sip wrote: Everyone seems to be working on their own servers that are in there homes, offices and elsewhere. The only common thread is this list-serv. I have several * servers set up here in Austin, Texas. I propose we set up one of them with with all the list-serv members. At this time the calls would be limited to on-net calls only to other members. Isn't this exactly what IAXTEL was supposed to be all about? or are you proposing somthing else? As the project grows and other members add servers we will decide if we want to give all servers PSTN access. Billing would be a major issue wih this, but its a cool idea.. This will give everyone a realtime enviroment in which to test their ideas, as well as being able to talk directly to other members if walk through TECH SUPPORT is needed. Sounds good.. ALSO, we can show Martin and Mark that tech support doesn't have to be in AL. I have to say the Digium tech support is pretty good.. I mailed in a query at 16:30 yesterday, AL time, and got a responce within about an hour.. So Let's TALK ABOUT IT!!! Yes.. lets.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This message was checked by MailScan for WorkgroupMail. www.workgroupmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
sip wrote: IAXTEL is a 1-700 system designed for on-net calls. We have very low-cost PSTN lines to 48 states for no-cost long-distance dialing. I plan to add some of these lines to the system. As servers are added in other cities around the world I envision dialing Berlin, Germany from Austin, Texas and using a local hop-off in Berlin to make a local call FREE. (No 2 cents per minute) If we could blend IAXTEL into our PSTN service that would be one major hurtle overcome. But that would put a programing burden on the guys at Digium and they seem to have their plate full. So Let's TALK ABOUT IT!!! I think Asterisk would need some new features before this would become a reality.. Basically it needs the voice equivilent of what RIP does for IP.. this would allow all servers in the network to know what extensions are avaliable on all other servers and the best way to get there.. Coupled with that it would need a shortest path routing system so that it would find the least cost route from UA1 to UA2 through a network of servers.. Without these you will have to have a central server with all the logic hard coded onto it, sufficent bandwidth to maintain the data rates for the peak symiltanious calls and enough processing power to handle encoding decoading and switching all those voice channels.. Or have you thought of a way around these issues? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
That is right!!! We will enable the Voice-Mail Email-Mail function. That way if you call a member and he is not there you leave him a message...it is packaged in a wav file...then Emailed. The next time he checks his email he will have the voice mail also! - Original Message - From: Manoj K Gupta [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 04, 2003 9:24 AM Subject: Re: [Asterisk-Users] Let's TALK ABOUT IT!!! I welocme the idea.After all we are deploying and using a voice based technology so we can have something like Voice Mailing List where we will be having voice messages and realtime talks with other programmers and developers. I think this should set a precedent for other mailing list also. Rgds Manoj K Gupta - Original Message - From: sip [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 04, 2003 7:31 PM Subject: Re: [Asterisk-Users] Let's TALK ABOUT IT!!! IAXTEL is a 1-700 system designed for on-net calls. We have very low-cost PSTN lines to 48 states for no-cost long-distance dialing. I plan to add some of these lines to the system. As servers are added in other cities around the world I envision dialing Berlin, Germany from Austin, Texas and using a local hop-off in Berlin to make a local call FREE. (No 2 cents per minute) If we could blend IAXTEL into our PSTN service that would be one major hurtle overcome. But that would put a programing burden on the guys at Digium and they seem to have their plate full. So Let's TALK ABOUT IT!!! - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 04, 2003 8:10 AM Subject: Re: [Asterisk-Users] Let's TALK ABOUT IT!!! sip wrote: Everyone seems to be working on their own servers that are in there homes, offices and elsewhere. The only common thread is this list-serv. I have several * servers set up here in Austin, Texas. I propose we set up one of them with with all the list-serv members. At this time the calls would be limited to on-net calls only to other members. Isn't this exactly what IAXTEL was supposed to be all about? or are you proposing somthing else? As the project grows and other members add servers we will decide if we want to give all servers PSTN access. Billing would be a major issue wih this, but its a cool idea.. This will give everyone a realtime enviroment in which to test their ideas, as well as being able to talk directly to other members if walk through TECH SUPPORT is needed. Sounds good.. ALSO, we can show Martin and Mark that tech support doesn't have to be in AL. I have to say the Digium tech support is pretty good.. I mailed in a query at 16:30 yesterday, AL time, and got a responce within about an hour.. So Let's TALK ABOUT IT!!! Yes.. lets.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This message was checked by MailScan for WorkgroupMail. www.workgroupmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This message was checked by MailScan for WorkgroupMail. www.workgroupmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
Each server would update from a master SQL database at a predetermined time. This way all servers would be in sync. - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 04, 2003 9:55 AM Subject: Re: [Asterisk-Users] Let's TALK ABOUT IT!!! sip wrote: IAXTEL is a 1-700 system designed for on-net calls. We have very low-cost PSTN lines to 48 states for no-cost long-distance dialing. I plan to add some of these lines to the system. As servers are added in other cities around the world I envision dialing Berlin, Germany from Austin, Texas and using a local hop-off in Berlin to make a local call FREE. (No 2 cents per minute) If we could blend IAXTEL into our PSTN service that would be one major hurtle overcome. But that would put a programing burden on the guys at Digium and they seem to have their plate full. So Let's TALK ABOUT IT!!! I think Asterisk would need some new features before this would become a reality.. Basically it needs the voice equivilent of what RIP does for IP.. this would allow all servers in the network to know what extensions are avaliable on all other servers and the best way to get there.. Coupled with that it would need a shortest path routing system so that it would find the least cost route from UA1 to UA2 through a network of servers.. Without these you will have to have a central server with all the logic hard coded onto it, sufficent bandwidth to maintain the data rates for the peak symiltanious calls and enough processing power to handle encoding decoading and switching all those voice channels.. Or have you thought of a way around these issues? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This message was checked by MailScan for WorkgroupMail. www.workgroupmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
At 09:01 4-10-2003 -0500, you wrote: IAXTEL is a 1-700 system designed for on-net calls. We have very low-cost PSTN lines to 48 states for no-cost long-distance dialing. I plan to add some of these lines to the system. As servers are added in other cities around the world I envision dialing Berlin, Germany from Austin, Texas and using a local hop-off in Berlin to make a local call FREE. (No 2 cents per minute) You are assuming the hop-off in berlin can be made 'free', but thats wrong. For most countries in europe local calls are _not_ free at all, so as mentioned, the billing will become an instantaneous issue... Its a nice idea to consider nontheless.. Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
sip wrote: Each server would update from a master SQL database at a predetermined time. This way all servers would be in sync. Ok I get it, so there is going to be some app or script that will update the DB from a central source, and an AGI written for Asterisk that will do the lookup in the DB that was synced to find the extension and the server that extension is connected to, and then initiate a connection directly with that server.. Sounds workable.. Now what about these free local PSTN calls?? not all countries have free local calls.. how do you think the billing will be organised? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
But it would be a free call to the common man who had a fast internet connection and a softphone or IP phone. He doesn't have to have a server or know the tech stuff... just a free softphone and he is in. After all, we are all working to develop this industry...build servers...sell phones...etc. - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 04, 2003 9:55 AM Subject: Re: [Asterisk-Users] Let's TALK ABOUT IT!!! At 09:01 4-10-2003 -0500, you wrote: IAXTEL is a 1-700 system designed for on-net calls. We have very low-cost PSTN lines to 48 states for no-cost long-distance dialing. I plan to add some of these lines to the system. As servers are added in other cities around the world I envision dialing Berlin, Germany from Austin, Texas and using a local hop-off in Berlin to make a local call FREE. (No 2 cents per minute) You are assuming the hop-off in berlin can be made 'free', but thats wrong. For most countries in europe local calls are _not_ free at all, so as mentioned, the billing will become an instantaneous issue... Its a nice idea to consider nontheless.. Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This message was checked by MailScan for WorkgroupMail. www.workgroupmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
Each server would update from a master SQL database at a predetermined time. This way all servers would be in sync. Ok I get it, so there is going to be some app or script that will update the DB from a central source, and an AGI written for Asterisk that will do the lookup in the DB that was synced to find the extension and the server that extension is connected to, and then initiate a connection directly with that server.. Actually, if this was to be done, it might be an idea to do it with DNS, so client machines would just do Dial(IAX2/[EMAIL PROTECTED]/442071234567) and the DNS system would resolve which machine is the correct target - no cleverness at all required at the client end, so implementation would be portable across all the other gnophones etc. Sounds workable.. Now what about these free local PSTN calls?? not all countries have free local calls.. how do you think the billing will be organised? However, this is a huge problem and I can't see it working.! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
Everyone seems to be working on their own servers that are in there homes, offices and elsewhere. The only common thread is this list-serv. I have several * servers set up here in Austin, Texas. I propose we set up one of them with with all the list-serv members. At this time the calls would be limited to on-net calls only to other members. As the project grows and other members add servers we will decide if we want to give all servers PSTN access. This will give everyone a realtime enviroment in which to test their ideas, as well as being able to talk directly to other members if walk through TECH SUPPORT is needed. From a high level perspective, don't many of us already have that? Seems the piece that's missing is a directory of those that would be willing to act in some sort of consulting/educational role since many of the list members have full time jobs elsewhere that may not be directly associated with *. (I'm purposefully not going to address the pstn access since many already have.) Why not add an Article to the www.voip-info.org site, and those that are interested with helping can list their FWD, IAXTEL, or other access number, probable hours of availability, any special focus skills, size of their current * environment, etc? I'm game. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
This entire idea will only work for the USA (and the few other countries that have unmetered local calls). HOWEVER, in most countries ALL calls, even if you call the person across the street are billed by the min. --Eric On Sat, 2003-10-04 at 10:33, sip wrote: But it would be a free call to the common man who had a fast internet connection and a softphone or IP phone. He doesn't have to have a server or know the tech stuff... just a free softphone and he is in. After all, we are all working to develop this industry...build servers...sell phones...etc. - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 04, 2003 9:55 AM Subject: Re: [Asterisk-Users] Let's TALK ABOUT IT!!! At 09:01 4-10-2003 -0500, you wrote: IAXTEL is a 1-700 system designed for on-net calls. We have very low-cost PSTN lines to 48 states for no-cost long-distance dialing. I plan to add some of these lines to the system. As servers are added in other cities around the world I envision dialing Berlin, Germany from Austin, Texas and using a local hop-off in Berlin to make a local call FREE. (No 2 cents per minute) You are assuming the hop-off in berlin can be made 'free', but thats wrong. For most countries in europe local calls are _not_ free at all, so as mentioned, the billing will become an instantaneous issue... Its a nice idea to consider nontheless.. Best regards, Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This message was checked by MailScan for WorkgroupMail. www.workgroupmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
Actually, if this was to be done, it might be an idea to do it with DNS, so client machines would just do Dial(IAX2/[EMAIL PROTECTED]/442071234567) and the DNS system would resolve which machine is the correct target - no cleverness at all required at the client end, so implementation would be portable across all the other gnophones etc. Yup. That would be the way to do it. I'll contribute the DNS code for it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
