[asterisk-users] How to add prefix in Extensions.Conf
Hello, I have a DID number 5672531308 , I want to add 92 prefix in it as been told by my provider , so I can I do this in extensions.conf? -- Regards, Muhammad Ali DIDx SUPPORT http://www.didx.net Skype: didxnet Phone: +1-212-655-5763 / +1-850-433-8555 Direct : +1-567-2531308 http://www.youtube.com/watch?v=acCDzP-7oqYfeature=player_embedded -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nat=yes
Hi, When NAT = YES, Asterisk server will extract IP from the network layer. When Nat = No, the Asterisk server will respond to the IP in the SIP header. Am I right? May be such type of options can be helpful for SIP application developers. Can't think of a scenario but If it is set to be YES for all peers, what will happen is that the response to all the SIP request will be routed to the IP in the network layer. IP's in the SIP header will be ignored, should not create a problem. Regards --- On Sun, 4/24/11, Steve Totaro stot...@asteriskhelpdesk.com wrote: From: Steve Totaro stot...@asteriskhelpdesk.com Subject: Re: [asterisk-users] Nat=yes To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sunday, April 24, 2011, 2:13 PM On Thu, Apr 21, 2011 at 5:42 AM, Alexandru Oniciuc alexandru.onic...@trivenet.it wrote: Dear * users, in your opinion, when using a * as a public server, is good practice enabling nat=yes in sip.conf for all the peers? Can anyone imagine a scenario when enabling this parameter (even for peers that don’t require it) can cause problems? Regards and thanks in advance,Alex I asked this same exact question several years ago. There are many replies with different takes. I would skim through Alex's posts, there is really nothing worth reading except it will break the SIP RFC handed down by the internets themselves. I use nat=yes all the time and it works just fine. http://www.mail-archive.com/asterisk-users@lists.digium.com/msg213941.html Nobody actually answered the question about the bad side, they just argued about the SIP RFC. Many others agreed to make it default behavior and that setting nat=yes gives a an extra degree of security. RFCs are great and all, but in the real world, phones just need to work. Thanks, Steve Totaro -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nat=yes
Hi, I am unsure of what you are saying. Just for discussion, if one has a control on the insertion of the IP address in the SIP header, then nat options working can be verified observed. In the OSI reference model, the Network is layer 3, IP. Call it Network, layer 3, or IP, it is the same. All-right, by IP from the network layer I meant, the IP address in the IP Header/Network layer/layer 3. IP from SIP I meant, SIP request generator's IP address in the SIP Header. I missed the word address. My customers don't really care for things that don't work. May be its useful for SIP application developers rather then end customers. Have a good time. Regards --- On Sun, 4/24/11, Steve Totaro stot...@asteriskhelpdesk.com wrote: From: Steve Totaro stot...@asteriskhelpdesk.com Subject: Re: [asterisk-users] Nat=yes To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sunday, April 24, 2011, 3:28 PM On Sun, Apr 24, 2011 at 5:55 AM, Muhammad Ali ali_...@yahoo.com wrote: Hi, When NAT = YES, Asterisk server will extract IP from the network layer. When Nat = No, the Asterisk server will respond to the IP in the SIP header. Am I right? May be such type of options can be helpful for SIP application developers. Can't think of a scenario but If it is set to be YES for all peers, what will happen is that the response to all the SIP request will be routed to the IP in the network layer. IP's in the SIP header will be ignored, should not create a problem. Regards --- On Sun, 4/24/11, Steve Totaro stot...@asteriskhelpdesk.com wrote: I am unsure of what you are saying. All I know is that setting nat=yes has never failed me when nat=no has and we are talking countless phones and installs. In the OSI reference model, the Network is layer 3, IP. Call it Network, layer 3, or IP, it is the same. nat=yes breaks the RFC due to NAT but it gets people talking. My customers don't really care for things that don't work. Thanks, Steve Totaro -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help Required---Problem in Installation without dahdi
Hi, Installation of dahdi requires kernel source that is not available with my remote virtual machine. Therefore I installed Asterisk without installing dahdi but when I start Asterisk it crashes while loading chan_agent.so (noload is also not useful in this case). Any suggestions or hints to overcome this issue? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
