Hi, I Work at an ISP and we have an Asterisk PBX. Our PBX doesn't have anything special, it just do the common things like auto attendant, CDR, call transfer and such things. We have 2 digium FXO cards to connect to the PSTN. The problem is that asterisk is not working very well, sometimes some of the dahdi channels get stuck and we need to release it through the asterisk console, the detection of callerid doesn't work (We live in brazil, dtmf signaling), and other detais that if I list all of them here it's going to take some time. I read a lot about asterisk and its bad design, I think that it added a lot a features without worrying about making stable implementations. My question is, can freeswitch fully replace our Asterisk PBX? Or it's not its purpose? Regards.
Hi Antonio, Yes, If you are not using a special feauture or module devolped for asterisk, You can replace your Asterisk with Freeswitch. As you said, you are in Brazil, FS support digium cards, also if I am not wrong also support khomp cards (Brazilian supplier). All you need is a small machine to give it a FS a try. Only two points: first,about fxo channels gettings stuck, this happens in several manufacture hardware when pstn lines are analog, it is better digital lines. You should take time in see if your digium hardware need some tunning like adjust tone detection, or line voltage. Second, abour your callerid, you need check with your pstn provider to validate. On 5/4/2012 8:35 AM, Antonio Modesto wrote:
Hi,
I Work at an ISP and we have an Asterisk PBX. Our PBX doesn't have anything special, it just do the common things like auto attendant, CDR, call transfer and such things. We have 2 digium FXO cards to connect to the PSTN. The problem is that asterisk is not working very well, sometimes some of the dahdi channels get stuck and we need to release it through the asterisk console, the detection of callerid doesn't work (We live in brazil, dtmf signaling), and other detais that if I list all of them here it's going to take some time. I read a lot about asterisk and its bad design, I think that it added a lot a features without worrying about making stable implementations. My question is, can freeswitch fully replace our Asterisk PBX? Or it's not its purpose?
Regards.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
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Hello Antonio, Your scenario is not uncommon. It may be that you have a particularly bad version of Asterisk. (A few releases were particularly problematic.) It could be that your hardware has issues. Or it could be that Asterisk may not be a good fit for your scenario. FreeSWITCH can do basically all of what you are doing now. The only thing I would do is contact Sangoma and ask them about Digium FXO cards in Brazil. (Sangoma wrote the FreeTDM stack for FreeSWITCH, so they're the experts on the subject.) If you are looking for a FreeSWITCH + GUI solution then you might want to check out FusionPBX or blue.box. -MC On Fri, May 4, 2012 at 6:05 AM, Antonio Modesto <modesto@isimples.com.br>wrote:
Hi,
I Work at an ISP and we have an Asterisk PBX. Our PBX doesn't have anything special, it just do the common things like auto attendant, CDR, call transfer and such things. We have 2 digium FXO cards to connect to the PSTN. The problem is that asterisk is not working very well, sometimes some of the dahdi channels get stuck and we need to release it through the asterisk console, the detection of callerid doesn't work (We live in brazil, dtmf signaling), and other detais that if I list all of them here it's going to take some time. I read a lot about asterisk and its bad design, I think that it added a lot a features without worrying about making stable implementations. My question is, can freeswitch fully replace our Asterisk PBX? Or it's not its purpose?
Regards.
Darío, thank you for remembering Khomp! Antonio, I might be able to chime in here. The CallerID is probably not working in Brazil because we use a specific protocol called DTMF. Most US hard card developers have a hard time with that because the standard in the US is FSK (a different protocol). Khomp has DTMF signaling support for both FXO and FXS cards (meaning your regular analog phones with display can use "bina" as well). Send me an email if you would like for more info. We support FreeSWITCH and Asterisk, for that matter. Regards (Abraços), -- João Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Friday, May 4, 2012 at 1:57 PM, Michael Collins wrote:
Hello Antonio,
Your scenario is not uncommon. It may be that you have a particularly bad version of Asterisk. (A few releases were particularly problematic.) It could be that your hardware has issues. Or it could be that Asterisk may not be a good fit for your scenario.
FreeSWITCH can do basically all of what you are doing now. The only thing I would do is contact Sangoma and ask them about Digium FXO cards in Brazil. (Sangoma wrote the FreeTDM stack for FreeSWITCH, so they're the experts on the subject.)
If you are looking for a FreeSWITCH + GUI solution then you might want to check out FusionPBX or blue.box.
-MC
On Fri, May 4, 2012 at 6:05 AM, Antonio Modesto <modesto@isimples.com.br (mailto:modesto@isimples.com.br)> wrote:
Hi,
I Work at an ISP and we have an Asterisk PBX. Our PBX doesn't have anything special, it just do the common things like auto attendant, CDR, call transfer and such things. We have 2 digium FXO cards to connect to the PSTN. The problem is that asterisk is not working very well, sometimes some of the dahdi channels get stuck and we need to release it through the asterisk console, the detection of callerid doesn't work (We live in brazil, dtmf signaling), and other detais that if I list all of them here it's going to take some time. I read a lot about asterisk and its bad design, I think that it added a lot a features without worrying about making stable implementations. My question is, can freeswitch fully replace our Asterisk PBX? Or it's not its purpose?
Regards.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org (mailto:consulting@freeswitch.org) http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Hi, It's good to know that the list is very active! Talking about the DTMF signaling, some guys on the Asterisk list told me that they got it running here in Brazil using DTMF. When I tried to configure it on my asterisk, I had a lot of erros, sometimes the caller id worked but it had some digits in inverted positions (It sounds very strange). As my asterisk is in production, I have a limited time to try this kind of thing on it, I will do some tests with FS in a free machine here. João, I liked to know that are active br guys in this list, sure I will contact you to exchange some ideas. Thank you very much. On Fri, 2012-05-04 at 17:49 -0300, João Mesquita wrote:
Darío, thank you for remembering Khomp!
Antonio, I might be able to chime in here. The CallerID is probably not working in Brazil because we use a specific protocol called DTMF. Most US hard card developers have a hard time with that because the standard in the US is FSK (a different protocol). Khomp has DTMF signaling support for both FXO and FXS cards (meaning your regular analog phones with display can use "bina" as well).
Send me an email if you would like for more info. We support FreeSWITCH and Asterisk, for that matter.
Regards (Abraços),
-- João Mesquita Sent with Sparrow
On Friday, May 4, 2012 at 1:57 PM, Michael Collins wrote:
Hello Antonio,
Your scenario is not uncommon. It may be that you have a particularly bad version of Asterisk. (A few releases were particularly problematic.) It could be that your hardware has issues. Or it could be that Asterisk may not be a good fit for your scenario.
FreeSWITCH can do basically all of what you are doing now. The only thing I would do is contact Sangoma and ask them about Digium FXO cards in Brazil. (Sangoma wrote the FreeTDM stack for FreeSWITCH, so they're the experts on the subject.)
If you are looking for a FreeSWITCH + GUI solution then you might want to check out FusionPBX or blue.box.
-MC
On Fri, May 4, 2012 at 6:05 AM, Antonio Modesto <modesto@isimples.com.br> wrote:
Hi,
I Work at an ISP and we have an Asterisk PBX. Our PBX doesn't have anything special, it just do the common things like auto attendant, CDR, call transfer and such things. We have 2 digium FXO cards to connect to the PSTN. The problem is that asterisk is not working very well, sometimes some of the dahdi channels get stuck and we need to release it through the asterisk console, the detection of callerid doesn't work (We live in brazil, dtmf signaling), and other detais that if I list all of them here it's going to take some time. I read a lot about asterisk and its bad design, I think that it added a lot a features without worrying about making stable implementations. My question is, can freeswitch fully replace our Asterisk PBX? Or it's not its purpose?
Regards.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- Atenciosamente, Antônio Modesto Gerente de TI Praça Getúlio Vargas, 77 – Sala 308 – Centro Santo Antônio do Monte – MG – CEP: 35560-000 Tel:(37) 3281-2800 Contato: isimples@isimples.com.br http://www.isimples.com.br Aviso:Esta mensagem e quaisquer arquivos em anexo podem conter informações confidenciais e/ou privilegiadas. Se você não for o destinatário ou a pessoa autorizada a receber esta mensagem, por favor, não leia, copie, repasse, imprima, guarde, nem tome qualquer ação baseada nessas informações. Notifique o remetente imediatamente por e-mail e apague a mensagem permanentemente. Atenção: embora a Isimples Telecom, tome seus cuidados para garantir a ausência de vírus neste e-mail, a empresa não se responsabiliza por quaisquer perdas ou danos decorrentes do uso da mensagem e seus anexos. A segurança e ausência de erros na transmissão do e-mail não podem ser garantidas, já que as informações podem ser interceptadas, corrompidas, perdidas, destruídas, atrasadas, chegarem incompletas, ou, ainda, conter vírus. Recomendamos checar se o e-mail e seus anexos contém vírus, uma vez que nem a Isimples Telecom ou o remetente se responsabilizam pela transmissão destes.
participants (4)
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Antonio Modesto
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João Mesquita
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Michael Collins
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Saugort Dario Garcia Tovar