Re: [asterisk-users] Connection question...
On Thu, Oct 12, 2006 at 01:50:53AM -0300, Danko Miocevic wrote: Dovid, the thing is that my server is on one internet connection connected directly... without nat... but my phones are behind a nat and they are in their own network, separated from the server. I just wanted to manage the calls with my server but making something like a direct connection between them... So do you mean something like this? phone 1 -|192.168.1.1 x.x.x.xy.y.y.y 192.168.1.200 +--- NAT --- Asterisk |F/W phone 2 --| 192.168.1.201 So the phones are pointing at Asterisk y.y.y.y as their SIP proxy, but if phone 1 places a call to phone 2 you want the audio to go directly between 192.168.1.200 and 192.168.1.201 on the LAN, is that it? You can get almost this if you install 'siproxd' on the NAT firewall, and configure the phones to use 192.168.1.1 as their outbound SIP proxy. What actually happens for an internal call is that the media stream will go from 192.168.1.200 to 192.168.1.1 to 192.168.1.201, but at least it stays on the LAN. Asterisk will believe that the phones are at IP address x.x.x.x, and will see SDP messages giving a media endpoint of x.x.x.x, because that's what siproxd does. You should probably then set 'nat=no' on the Asterisk SIP channels, because Asterisk no longer needs to do any special NAT processing. siproxd doesn't handle multiple concurrent registrations of the same SIP user ID, but then neither does Asterisk yet, so that won't be a limitation for you. That's the simplest solution I know of. You can play with openser but that's a much more complex beastie, essentially a toolbox for forwarding and modifying SIP packets. Or if you're happier with Asterisk, you could install another Asterisk server on the NAT F/W or on the LAN. (But that's much more complex to setup and maintain than siproxd; with siproxd on the LAN all the call routing intelligence still resides on your main Asterisk server) I'm not 100% sure, but I don't think your external Asterisk server can do what you want without assistance from a local SIP proxy on the LAN. Setting 'canreinvite=yes' would allow Asterisk to point the phones at each other, but you've probably already had to set 'nat=yes' which means that Asterisk ignores a lot of the private IP information contained within the SIP/SDP packets and uses the source IP address it sees (x.x.x.x) instead. HTH, Brian. ___ --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] Connection question...
Dovid, the thing is that my server is on one internet connection connected directly... without nat... but my phones are behind a nat and they are in their own network, separated from the server. I just wanted to manage the calls with my server but making something like a direct connection between them... I don´t speak really good english.. do you know what I mean? Danko - Original Message - From: Dovid B [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 10, 2006 5:28 PM Subject: Re: [asterisk-users] Connection question... - Original Message - From: Danko Miocevic [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, October 10, 2006 7:16 PM Subject: [asterisk-users] Connection question... I want to try something with my asterisk but I have something that I need to know. The thing is that I am behind a NAT (I have to phones in a lan connected to the internet with a router), my server is directly conected to the internet on a different connection (in another place). I make a call from one phone to the other, but will they connect directly inside my lan? will I need an important Internet connection (I mean fast)? what info will be transfered from the server to the phones and from the phones to the server? If someone know something about this, I will appreciate any info, thanks to all, Danko Danko, If your server is behind NAT then from what I know the RTP stream must go thru the server. Regarding the kind of internet connection it all depends on how many concurent calls as well as what kind of codecs you are using. If you want to share your current internet with the phones then make sure to set up GOOD QoS. ___ --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
[asterisk-users] Connection question...
I want to try something with my asterisk but I have something that I need to know. The thing is that I am behind a NAT (I have to phones in a lan connected to the internet with a router), my server is directly conected to the internet on a different connection (in another place). I make a call from one phone to the other, but will they connect directly inside my lan? will I need an important Internet connection (I mean fast)? what info will be transfered from the server to the phones and from the phones to the server? If someone know something about this, I will appreciate any info, thanks to all, Danko ___ --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] Connection question...
- Original Message - From: Danko Miocevic [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, October 10, 2006 7:16 PM Subject: [asterisk-users] Connection question... I want to try something with my asterisk but I have something that I need to know. The thing is that I am behind a NAT (I have to phones in a lan connected to the internet with a router), my server is directly conected to the internet on a different connection (in another place). I make a call from one phone to the other, but will they connect directly inside my lan? will I need an important Internet connection (I mean fast)? what info will be transfered from the server to the phones and from the phones to the server? If someone know something about this, I will appreciate any info, thanks to all, Danko Danko, If your server is behind NAT then from what I know the RTP stream must go thru the server. Regarding the kind of internet connection it all depends on how many concurent calls as well as what kind of codecs you are using. If you want to share your current internet with the phones then make sure to set up GOOD QoS. ___ --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] Connection question
As far as I know you can. The only thing you need to know is what ports does your Alcatel PBX use. Tomislav Asterisk seems to be a very good peace of software, but i am interested to know if i can use plain ISDN cards with it, i mean use the isdn cards as a passthrough device between my alcatel pbx and voip users. thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connection question
Asterisk seems to be a very good peace of software, but i am interested to know if i can use plain ISDN cards with it, i mean use the isdn cards as a passthrough device between my alcatel pbx and voip users. thanks DLS - Projectos, Automação e Manutenção, Lda. João Carneiro, Tecnico Dep. Sistemas de Informação Rua da Boavista S/N - P.O.Box 313 4416-901 Grijó www.dls.pt Email: [EMAIL PROTECTED] Tel : +351 227 470 786 Fax : +351 227 470 787 Tlm : Esta mensagem de correio electrónico e qualquer dos seus ficheiros anexos, caso existam, são confidenciais e destinados apenas à(s) pessoa(s) ou entidade(s) acima referida(s), podendo conter informação confidencial, privilegiada, a qual não devera ser divulgada, copiada, gravada ou distribuida nos termos da lei vigente. Se não é o destinatário da mensagem, ou se ela lhe foi enviada por engano, agradecemos que não faça uso ou divulgação da mesma. A distribuição ou utilização da informação nela contida é VEDADA. Se recebeu esta mensagem por engano, por favor avise-nos de imediato, por correio electrónico, para o endereço acima e apague este e-mail do seu sistema. Obrigado This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
