Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Ok, on peut parler français alors ;) Olivier Jean-Michel Hiver a écrit : Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño, y si ayuda mucho y molesta poco, ¿por qué quejarse? Quel bordel, sacrebleu! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
How many channels have you guys been able to get with this? The only problem I have with this is that it takes skype and a soundcard (virtual or otherwise) and the API is really executing commands on a running skype process. In my opinion its not worth it for 1 concurrent call per account. I have written code that works with skype in linux that simulates a virtual sound device. I have used that and successfully done calls out with this. I havent played with the dbus stuff (how you control the skype app from within linux) but since I have a soundcard that I know the audio format of it wouldnt be difficult to integrate this into asterisk, I could tweak chan_oss and make it into chan_skype fairly easily since that takes care of the other half of the equation. The only thing missing would be the events via dbus, which there are plenty of examples on so its not like all new code would have to be written. But its just not worth it if you have to have skype running for each call. And then you would potentially have to have a new username for each running process, and skype really wants X on linux so you would have to at least have the X virtual frame buffer (it works and acts like X but never displays anything or uses any hardware). That seems like an aweful lot of wasted resources on a box to connect to skype. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Well, look at it this way: if you get the working, you can buy one of those tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia soundcard and a ethernet port. Run Linux off a CF card and have it setup to *only* interface with Skype and Asterisk. Basically, make a Skype ATA, but it would convert Skype to SIP. I think that could still be considered an ATA, right? Or a gateway at least. Since you can make a Skype account for free and can (for right now) make US and Canada LD calls for free, I think the cost and time to make them would be worth it. :) And if you figure out a good price for them, people might even buy them from you Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: How many channels have you guys been able to get with this? The only problem I have with this is that it takes skype and a soundcard (virtual or otherwise) and the API is really executing commands on a running skype process. In my opinion its not worth it for 1 concurrent call per account. I have written code that works with skype in linux that simulates a virtual sound device. I have used that and successfully done calls out with this. I havent played with the dbus stuff (how you control the skype app from within linux) but since I have a soundcard that I know the audio format of it wouldnt be difficult to integrate this into asterisk, I could tweak chan_oss and make it into chan_skype fairly easily since that takes care of the other half of the equation. The only thing missing would be the events via dbus, which there are plenty of examples on so its not like all new code would have to be written. But its just not worth it if you have to have skype running for each call. And then you would potentially have to have a new username for each running process, and skype really wants X on linux so you would have to at least have the X virtual frame buffer (it works and acts like X but never displays anything or uses any hardware). That seems like an aweful lot of wasted resources on a box to connect to skype. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461 DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Wed, June 28, 2006 10:14, [EMAIL PROTECTED] said: Well, look at it this way: if you get the working, you can buy one of those tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia soundcard and a ethernet port. Run Linux off a CF card and have it setup to *only* interface with Skype and Asterisk. Basically, make a Skype ATA, but it would convert Skype to SIP. I think that could still be considered an ATA, right? Or a gateway at least. Since you can make a Skype account for free and can (for right now) make US and Canada LD calls for free, I think the cost and time to make them would be worth it. :) And if you figure out a good price for them, people might even buy them from you Undrhil Another advantage is that you can reach all those people who have Skype and are not willing to try Voipbuster or similar SIP based providers, and tell them that SIP/IAX/Asterisk *is* the better solution, because they cannot do the same with Skype the other way round! ;-p -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Wed, 2006-06-28 at 08:14 +, [EMAIL PROTECTED] wrote: Well, look at it this way: if you get the working, you can buy one of those tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia soundcard and a ethernet port. Run Linux off a CF card and have it setup to *only* interface with Skype and Asterisk. Basically, make a Skype ATA, but it would convert Skype to SIP. I think that could still be considered an ATA, right? Or a gateway at least. it wouldnt need a real soundcard, that is part of the point. I remap all the calls the same way that I did for allowing instant porting of your digium g729 licenses (in another post, code is at my personal site http://www.0xdecafbad.com/ somewhere). Remapping those calls is trivial, there are very few things that are acutally done to a soundcard to set it up, ioctl() for setting the sample rate, etc and read/write/open/close basically. Really trivial code. It would however be nicer if you didnt have to run a seperate copy of the binary for each call. This has a direct cost against memory. It would be better if it didnt use memory to open a GUI (even with the virtual framebuffer for X it still takes all that memory even though it doesnt display for real). I also doubt that a 386 would cut it, with everything going on it would have to be faster and that pushes the cost up. If you are going to do that it might be cheaper to buy one of the 1,2,4 port FSX/FXO devices for integrating with a phone system or something (some plug into wall jacks others into phones). The 4 ports are about $750 which is steep. The 99 port one which is unclear how you use it exactly is $1500 or so. Actualy looking at the 99 port model it appears that its just a usb soundcard that has a FXS port on it, which is a silly way in my opinion, and still requires a system running skype to work :( Since you can make a Skype account for free and can (for right now) make US and Canada LD calls for free, I think the cost and time to make them would be worth it. :) And if you figure out a good price for them, people might even buy them from you I dont, the overhead is insane. As as for a price for 'them' it would just be a software program. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Wed, Jun 28, 2006 at 08:14:56AM -, [EMAIL PROTECTED] wrote: Well, look at it this way: if you get the working, you can buy one of those tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia soundcard and a ethernet port. Run Linux off a CF card and have it setup to *only* interface with Skype and Asterisk. Basically, make a Skype ATA, but it would convert Skype to SIP. I think that could still be considered an ATA, right? Or a gateway at least. Since you can make a Skype account for free and can (for right now) make US and Canada LD calls for free, I think the cost and time to make them would be worth it. :) And if you figure out a good price for them, people might even buy them from you You would be violating the terms of usage of their API if you want to use (let alone sell) such a device. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Wed, 2006-06-28 at 13:15 +0300, Tzafrir Cohen wrote: Since you can make a Skype account for free and can (for right now) make US and Canada LD calls for free, I think the cost and time to make them would be worth it. :) And if you figure out a good price for them, people might even buy them from you You would be violating the terms of usage of their API if you want to use (let alone sell) such a device. I am unsure if all the hardware devices are basically usb soundcards or not, havent really looked, but if they arent then it would seem to me that its possible to do. Further I dont think it would be against their api to write sofeware that uses their api. That is what was being discussed when this comment came out, so ... -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Hi, Is it illegal to use Uplink Skype2Sip software to connect a skype account to a homepbx asterisk? ( Just to know... i don't want to be bored because of asteriskpt.blogspot) On 6/28/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Wed, 2006-06-28 at 13:15 +0300, Tzafrir Cohen wrote: Since you can make a Skype account for free and can (for right now) make US and Canada LD calls for free, I think the cost and time to make them would be worth it. :) And if you figure out a good price for them, people might even buy them from you You would be violating the terms of usage of their API if you want to use (let alone sell) such a device. I am unsure if all the hardware devices are basically usb soundcards or not, havent really looked, but if they arent then it would seem to me that its possible to do. Further I dont think it would be against their api to write sofeware that uses their api. That is what was being discussed when this comment came out, so ... -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (GNU/Linux) iD8DBQBEol9n+1olxlzQw5cRAvmYAJ463UBN/3F1bkCo3smt92QaQhPzOACfSn/j OijC0wHuU8hmynUp/Osa6gA= =hEQW -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
You have all our respect. At least mine. Carry on! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
I agree whole-heartedly. If I could run this on my dedicated Asterisk machine it would be perfect... On 6/28/06, Matthias Fechner [EMAIL PROTECTED] wrote: Hi Marco, Marco Mouta wrote: Please feel free to contact me if you have more ideas to improve this solution, currently i didn't test more than one simultaneous calls incoming and outgoing through Skype. get it running on unix so you can run it on the asterisk server. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? Saudações Josué 2006/6/26, John Klimek [EMAIL PROTECTED]: I agree whole-heartedly.If I could run this on my dedicated Asteriskmachine it would be perfect... On 6/28/06, Matthias Fechner [EMAIL PROTECTED] wrote: Hi Marco, Marco Mouta wrote: Please feel free to contact me if you have more ideas to improve this solution, currently i didn't test more than one simultaneous calls incoming and outgoing through Skype. get it running on unix so you can run it on the asterisk server. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Bom dia, On 6/26/06, Josué Conti [EMAIL PROTECTED] wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? Sim é um software da Uplink, disponível para download gratuitamente, n garanto q seja freeware (talvez tenha limitações esta versao free) Podes ver a demo no site: http://asteriskpt.blogspot.com Se tudo estiver ok, deverás ouvir Musiconhold de um asterisk q tenho em casa ( n estou la agora). É freeware? Podemos seguir com o projeto Asterisk-PT? Claro que sim! http://asteriskpt.blogspot.com Podes por posts la, vou criar contas para podermos cooperar no blog. Se preferirem um site ou outra solução, estou aberto a sugestões. Saudações Josué 2006/6/26, John Klimek [EMAIL PROTECTED]: I agree whole-heartedly. If I could run this on my dedicated Asterisk machine it would be perfect... On 6/28/06, Matthias Fechner [EMAIL PROTECTED] wrote: Hi Marco, Marco Mouta wrote: Please feel free to contact me if you have more ideas to improve this solution, currently i didn't test more than one simultaneous calls incoming and outgoing through Skype. get it running on unix so you can run it on the asterisk server. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
OK Marco, irei efetuar os testes. Se você quiser, posso lhe ajudar no forum, estou a disposição. Assim que você criar as contas avise para podermos já ir colaborando. Saudações Josué 2006/6/26, Marco Mouta [EMAIL PROTECTED]: Bom dia,On 6/26/06, Josué Conti [EMAIL PROTECTED] wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo?Sim é um software da Uplink, disponível para download gratuitamente, n garanto q seja freeware (talvez tenha limitações esta versao free)Podes ver a demo no site:http://asteriskpt.blogspot.comSe tudo estiver ok, deverás ouvir Musiconhold de um asterisk q tenho em casa ( n estou la agora). É freeware? Podemos seguir com o projeto Asterisk-PT?Claro que sim! http://asteriskpt.blogspot.comPodes por posts la, vou criar contas para podermos cooperar no blog. Se preferirem um site ou outra solução, estou aberto a sugestões. Saudações Josué 2006/6/26, John Klimek [EMAIL PROTECTED] : I agree whole-heartedly.If I could run this on my dedicated Asterisk machine it would be perfect...On 6/28/06, Matthias Fechner [EMAIL PROTECTED] wrote: Hi Marco, Marco Mouta wrote:Please feel free to contact me if you have more ideas to improve thissolution, currently i didn't test more than one simultaneous calls incoming and outgoing through Skype. get it running on unix so you can run it on the asterisk server. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --Com os melhores cumprimentos,Marco Mouta___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. I don't know Portuguese and my Spanish is terrible, but I understood that Josue wanted to know if he needed any external modules. Marco pointed him to the right place to get skype-to-sip and now they're going to collaborate. So, please guys English please or you'll get more of my bad translations. ;) Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. Let them talk. What's it hurt the rest of us? We have seen the wages of tortured English sometimes unleashed on the list. If they're getting the job done, I say hit the Delete button and get on with your life. If 80% of the list traffic were in foreign languages, then I would say we would have an issue. MO. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Sorry to all, Now only English speaking :) Your translation was perfect. Thanks once more On 6/26/06, Mike Fedyk [EMAIL PROTECTED] wrote: Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. I don't know Portuguese and my Spanish is terrible, but I understood that Josue wanted to know if he needed any external modules. Marco pointed him to the right place to get skype-to-sip and now they're going to collaborate. So, please guys English please or you'll get more of my bad translations. ;) Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Sorry to all. Speaking English only. Regards Josué 2006/6/26, Marco Mouta [EMAIL PROTECTED]: Sorryto all,Now only English speaking :)Your translation was perfect.Thanks once more On 6/26/06, Mike Fedyk [EMAIL PROTECTED] wrote: Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. I don't know Portuguese and my Spanish is terrible, but I understood that Josue wanted to know if he needed any external modules.Marco pointed him to the right place to get skype-to-sip and now they're going to collaborate. So, please guys English please or you'll get more of my bad translations. ;) Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--Com os melhores cumprimentos,Marco Mouta___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Mon, June 26, 2006 20:06, Brian Capouch said: Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. Let them talk. What's it hurt the rest of us? It is more a question of netiquette... If you're on an English mailinglist, you should speak English (Not attacking Josué and Marco, just answering the question here). It is not only more productive (If you keep to English, more people understand and can contribute to *and* profit from the discussion), but speaking a different language not spoken by the majority on list is generally considered akin whispering in company: not quite rude, but also not-done... We have seen the wages of tortured English sometimes unleashed on the list. If they're getting the job done, I say hit the Delete button and get on with your life. You can hit the delete button for bad English too, you know! ;-) If 80% of the list traffic were in foreign languages, then I would say we would have an issue. Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Francesco Peeters (Asterisk) wrote: Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño, y si ayuda mucho y molesta poco, ¿por qué quejarse? B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On 6/26/06, Josué Conti [EMAIL PROTECTED] wrote: OK Marco, irei efetuar os testes. Se você quiser, posso lhe ajudar no forum, estou a disposição. Assim que você criar as contas avise para podermos já ir colaborando. Saudações JosuéThe differences of licenses are here: https://www.nch.com.au/cgi-bin/register.exe?software=uplink The site only says that support is different.-- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Mon, June 26, 2006 21:39, Brian Capouch said: Francesco Peeters (Asterisk) wrote: Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño, y si ayuda mucho y molesta poco, ¿por qué quejarse? B. Ningunas quejas aquí... Apenas una explicación en el 'netiquette' --FP ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Hi All. Please, we need to have more respect with the list. Regards Josué 2006/6/26, Francesco Peeters [EMAIL PROTECTED]: On Mon, June 26, 2006 21:39, Brian Capouch said: Francesco Peeters (Asterisk) wrote: Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten!;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño, y si ayuda mucho y molesta poco, ¿por qué quejarse? B.Ningunas quejas aquí... Apenas una explicación en el 'netiquette'--FP___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño, y si ayuda mucho y molesta poco, ¿por qué quejarse? Quel bordel, sacrebleu! -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Asterisk handling My Skype Calls This is for me, once more, Asterisk as the Future of Telephony. Today I've integrated my Skype Account as SIP extension in my * Box. This has been possible using Uplink Skype to SIP Adapter, available for free at http://www.nch.com.au/skypetosip/index.html . Main features that any one can easily integrate into Asterisk: - Route skype incoming calls into Asterisk DialPlan, then you just can do ANYThing route to your mobile, Meetme rooms, IVRs do it in your way. - Dialout calls from any SIP extension through Skype (reaching Skype contacts or outgoing calls to landline through Skype Outgoing calls prices. - Enable your website with SkypeMe Button and route it to Asterisk! Feel free to listen MusicOnHold from my Asterisk Box through my Skype Account. Check this in http://asteriskpt.blogspot.com - AsteriskPT - Asterisk Portuguese Users Group. Please feel free to contact me if you have more ideas to improve this solution, currently i didn't test more than one simultaneous calls incoming and outgoing through Skype. MoutaPT http://asteriskpt.blogspot.com - AsteriskPT - Asterisk Portuguese Users Group. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
Hi Marco, Marco Mouta wrote: Please feel free to contact me if you have more ideas to improve this solution, currently i didn't test more than one simultaneous calls incoming and outgoing through Skype. get it running on unix so you can run it on the asterisk server. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
