[asterisk-users] NAT yes
Hello averybody, In a no natted environment if I letnat=yes on sip.conf it would cause some thing bad or it is irrelevant ? Anybody know ? thanks in advanced! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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
On 07/26/2011 09:19 AM, Flavio Miranda wrote: In a no natted environment if I letnat=yes on sip.conf it would cause some thing bad or it is irrelevant ? Anybody know ? There is no harm unless the endpoint you are dealing with does not do symmetric RTP. The nat=yes option assumes that it is okay to send RTP back to the source port from which it originated, irrespectively of what's in the SDP. This will cause one-way audio if the endpoint happens to want to receive RTP on a different port than the one it is sending it from. Almost all endpoints these days do symmetric RTP, though, so it's not a huge concern. That said, from a methodological and aesthetic perspective, it is better not to break standard RFC-compliant behaviour unnecessarily. Thus, I would not enable nat=yes unless there really is no direct network and transport-layer reachability to the endpoint. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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
Thanks Alex Balashov, I am experiencing some one-way audio, that's the reason of the questions! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Tue, 26 Jul 2011 09:23:42 -0400 From: abalas...@evaristesys.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] NAT yes On 07/26/2011 09:19 AM, Flavio Miranda wrote: In a no natted environment if I letnat=yes on sip.conf it would cause some thing bad or it is irrelevant ? Anybody know ? There is no harm unless the endpoint you are dealing with does not do symmetric RTP. The nat=yes option assumes that it is okay to send RTP back to the source port from which it originated, irrespectively of what's in the SDP. This will cause one-way audio if the endpoint happens to want to receive RTP on a different port than the one it is sending it from. Almost all endpoints these days do symmetric RTP, though, so it's not a huge concern. That said, from a methodological and aesthetic perspective, it is better not to break standard RFC-compliant behaviour unnecessarily. Thus, I would not enable nat=yes unless there really is no direct network and transport-layer reachability to the endpoint. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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
On 07/26/2011 09:29 AM, Flavio Miranda wrote: I am experiencing some one-way audio, that's the reason of the questions! There are many possible reasons for it, but asymmetric RTP + 'nat=yes' may be one of them. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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
Also consider the setting localnet in sip.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Tuesday, July 26, 2011 9:24 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] NAT yes On 07/26/2011 09:19 AM, Flavio Miranda wrote: In a no natted environment if I letnat=yes on sip.conf it would cause some thing bad or it is irrelevant ? Anybody know ? There is no harm unless the endpoint you are dealing with does not do symmetric RTP. The nat=yes option assumes that it is okay to send RTP back to the source port from which it originated, irrespectively of what's in the SDP. This will cause one-way audio if the endpoint happens to want to receive RTP on a different port than the one it is sending it from. Almost all endpoints these days do symmetric RTP, though, so it's not a huge concern. That said, from a methodological and aesthetic perspective, it is better not to break standard RFC-compliant behaviour unnecessarily. Thus, I would not enable nat=yes unless there really is no direct network and transport-layer reachability to the endpoint. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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
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 -- _ -- 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
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 -- _ -- 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
Re: [asterisk-users] Nat=yes
check this http://www.voip-info.org/wiki/view/Asterisk+sip+nat On Thu, Apr 21, 2011 at 2:12 PM, 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 -- _ -- 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
Dear Benjamin; So in that case, when we set nat = yes? For what we do this? C F, I have nat=yes set by default for all my extensions(with canreinvite=no). And things work fine. Bilal, about Asterisk sending packets to public/private : Asterisk will send packets to the public IP advertised by the msg/recv from address. It is the NAT's headache on the endpoints network periphery to send the response from Asterisk to the endpoint. C F wrote: If you set yes then asterisk assumes that the address its coming from is not the same as the UA thinks it is. most devices will not operate properly if set to yes when they are in fact local. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] nat=yes
So I'll rephrase to some devices will not operate properly, since after your message I am assuming that you tested this with most devices. On 9/10/07, Benjamin Jacob [EMAIL PROTECTED] wrote: C F, I have nat=yes set by default for all my extensions(with canreinvite=no). And things work fine. Bilal, about Asterisk sending packets to public/private : Asterisk will send packets to the public IP advertised by the msg/recv from address. It is the NAT's headache on the endpoints network periphery to send the response from Asterisk to the endpoint. C F wrote: If you set yes then asterisk assumes that the address its coming from is not the same as the UA thinks it is. most devices will not operate properly if set to yes when they are in fact local. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] nat=yes
C F wrote: BTW, AFAIK, there is no such thing as host=static it's either dynamic or an IP/Name. Yeah, I learned that the hard way. I had only set up dynamic devices for a couple of months, and the first time I had reason to set up a device with a static IP, I just assumed that 'host=static' would work in sip.conf. Dur, it took me a couple of hours to figure out why my fax machine could fax, but not receive. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] nat=yes
Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] nat=yes
If you set yes then asterisk assumes that the address its coming from is not the same as the UA thinks it is. most devices will not operate properly if set to yes when they are in fact local. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] nat=yes
BTW, AFAIK, there is no such thing as host=static it's either dynamic or an IP/Name. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] nat=yes
C F, I have nat=yes set by default for all my extensions(with canreinvite=no). And things work fine. Bilal, about Asterisk sending packets to public/private : Asterisk will send packets to the public IP advertised by the msg/recv from address. It is the NAT's headache on the endpoints network periphery to send the response from Asterisk to the endpoint. C F wrote: If you set yes then asterisk assumes that the address its coming from is not the same as the UA thinks it is. most devices will not operate properly if set to yes when they are in fact local. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] nat=yes and qualify=yes viable NAT solutions?
Looking for some feedback on whether nat=yes and qualify=yes will provide a workable solution in many cases? The * server is on a public address, no NAT, the UAs (sipura, linksys, polycom) are behind various types of NAT. Obviously port mapping in the NAT device works, but what about cases where there is no admin access possible to the NAT device? From what I have read nat=yes makes asterisk look at the udp header instead of the SIP info for the UA IP address, and qualify=yes sends keepalives in the form of frequent SIP OPTIONS queries. I think I also read that nat=yes enables symmetrical RTP What I do not know is does it work in the real world? ___ --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] nat=yes and qualify=yes viable NAT solutions?
Sipura works, I never tried linksys, Polycom might and might not work. On 2/26/06, Damon Estep [EMAIL PROTECTED] wrote: Looking for some feedback on whether nat=yes and qualify=yes will provide a workable solution in many cases? The * server is on a public address, no NAT, the UAs (sipura, linksys, polycom) are behind various types of NAT. Obviously port mapping in the NAT device works, but what about cases where there is no admin access possible to the NAT device? From what I have read nat=yes makes asterisk look at the udp header instead of the SIP info for the UA IP address, and qualify=yes sends keepalives in the form of frequent SIP OPTIONS queries. I think I also read that nat=yes enables symmetrical RTP What I do not know is does it work in the real world? ___ --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] nat=yes and qualify=yes viable NAT solutions?
Thanks, the linksys is a sipura, so what works on one should work on the other. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, February 26, 2006 7:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] nat=yes and qualify=yes viable NAT solutions? Sipura works, I never tried linksys, Polycom might and might not work. On 2/26/06, Damon Estep [EMAIL PROTECTED] wrote: Looking for some feedback on whether nat=yes and qualify=yes will provide a workable solution in many cases? The * server is on a public address, no NAT, the UAs (sipura, linksys, polycom) are behind various types of NAT. Obviously port mapping in the NAT device works, but what about cases where there is no admin access possible to the NAT device? From what I have read nat=yes makes asterisk look at the udp header instead of the SIP info for the UA IP address, and qualify=yes sends keepalives in the form of frequent SIP OPTIONS queries. I think I also read that nat=yes enables symmetrical RTP What I do not know is does it work in the real world? ___ --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
Re: [Asterisk-Users] nat=yes and qualify=yes viable NAT solutions?
On 2/26/06, Damon Estep [EMAIL PROTECTED] wrote: Thanks, the linksys is a sipura, so what works on one should work on the other. I wouldn't bet on it, since the SPA-xxx where out before the merge. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, February 26, 2006 7:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] nat=yes and qualify=yes viable NAT solutions? Sipura works, I never tried linksys, Polycom might and might not work. On 2/26/06, Damon Estep [EMAIL PROTECTED] wrote: Looking for some feedback on whether nat=yes and qualify=yes will provide a workable solution in many cases? The * server is on a public address, no NAT, the UAs (sipura, linksys, polycom) are behind various types of NAT. Obviously port mapping in the NAT device works, but what about cases where there is no admin access possible to the NAT device? From what I have read nat=yes makes asterisk look at the udp header instead of the SIP info for the UA IP address, and qualify=yes sends keepalives in the form of frequent SIP OPTIONS queries. I think I also read that nat=yes enables symmetrical RTP What I do not know is does it work in the real world? ___ --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 ___ --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] NAT=YES
Don't have a clue why changing your settings causes the phone call to fail, but obviously it needs to be investigated. The keyword is not supported in code, therefore something else is impacting your config. OK, so I have a nonexistant line in my settings. Why then when I remove it does my phone call fail? Rich Adamson wrote: FYI, there is no such thing as reinvite. Someone started using that in postings a long time ago and a few people keep posting it, but it doesn't exist. (Check /usr/src/astersik/configs/sip.conf.sample) Add canreinvite=no and reinvite=no to the relevant stanza in sip.conf Mark Klint, Peter wrote: Good morning Does anyone have experience with NAT=YES? I have the following configuration and am a bit confused as to why the Asterisk server initially sends out RTP to the remote host private IP and then switches to the public IP. Configuration Info: I have all users in SIP.CONF configured with NAT=YES Asterisk has a public IP Remote host is behind a firewall with NAT When I sniff on the Asterisk public network, I see the following. 1. INVITE from remote host public IP to Asterisk public IP 2. 183 response from Asterisk public IP to remote host public IP 3. RTP from Asterisk public IP to the remote host private IP 4. RTP from remote host public IP to Asterisk public IP 5. RTP from Asterisk public IP to the remote host public IP Is there a way to prevent step 3 from happening? Or, is there a way to delay the invalid RTP from being sent from the Asterisk in step 3? Does anyone know why the Asterisk sends RTP to remote host private IP? I would expect NAT=YES to correct this issue. Thanks, Peter ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 ---End of Original Message- ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 ---End of Original Message- ___ 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
Re: [Asterisk-Users] NAT=YES
canreinvite= is a valid option. reinvite= is not a valid option. Rich Adamson wrote: Don't have a clue why changing your settings causes the phone call to fail, but obviously it needs to be investigated. The keyword is not supported in code, therefore something else is impacting your config. OK, so I have a nonexistant line in my settings. Why then when I remove it does my phone call fail? Rich Adamson wrote: FYI, there is no such thing as reinvite. Someone started using that in postings a long time ago and a few people keep posting it, but it doesn't exist. (Check /usr/src/astersik/configs/sip.conf.sample) -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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] NAT=YES
Title: NAT=YES Good morning Does anyone have experience with NAT=YES? I have the following configuration and am a bit confused as to why the Asterisk server initially sends out RTP to the remote host private IP and then switches to the public IP. Configuration Info: I have all users in SIP.CONF configured with NAT=YES Asterisk has a public IP Remote host is behind a firewall with NAT When I sniff on the Asterisk public network, I see the following. 1. INVITE from remote host public IP to Asterisk public IP 2. 183 response from Asterisk public IP to remote host public IP 3. RTP from Asterisk public IP to the remote host private IP 4. RTP from remote host public IP to Asterisk public IP 5. RTP from Asterisk public IP to the remote host public IP Is there a way to prevent step 3 from happening? Or, is there a way to delay the invalid RTP from being sent from the Asterisk in step 3? Does anyone know why the Asterisk sends RTP to remote host private IP? I would expect NAT=YES to correct this issue. Thanks, Peter ___ 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
Re: [Asterisk-Users] NAT=YES
Add canreinvite=no and reinvite=no to the relevant stanza in sip.conf Mark Klint, Peter wrote: Good morning Does anyone have experience with NAT=YES? I have the following configuration and am a bit confused as to why the Asterisk server initially sends out RTP to the remote host private IP and then switches to the public IP. Configuration Info: I have all users in SIP.CONF configured with NAT=YES Asterisk has a public IP Remote host is behind a firewall with NAT When I sniff on the Asterisk public network, I see the following. 1. INVITE from remote host public IP to Asterisk public IP 2. 183 response from Asterisk public IP to remote host public IP 3. RTP from Asterisk public IP to the remote host private IP 4. RTP from remote host public IP to Asterisk public IP 5. RTP from Asterisk public IP to the remote host public IP Is there a way to prevent step 3 from happening? Or, is there a way to delay the invalid RTP from being sent from the Asterisk in step 3? Does anyone know why the Asterisk sends RTP to remote host private IP? I would expect NAT=YES to correct this issue. Thanks, Peter ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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
Re: [Asterisk-Users] NAT=YES
FYI, there is no such thing as reinvite. Someone started using that in postings a long time ago and a few people keep posting it, but it doesn't exist. (Check /usr/src/astersik/configs/sip.conf.sample) Add canreinvite=no and reinvite=no to the relevant stanza in sip.conf Mark Klint, Peter wrote: Good morning Does anyone have experience with NAT=YES? I have the following configuration and am a bit confused as to why the Asterisk server initially sends out RTP to the remote host private IP and then switches to the public IP. Configuration Info: I have all users in SIP.CONF configured with NAT=YES Asterisk has a public IP Remote host is behind a firewall with NAT When I sniff on the Asterisk public network, I see the following. 1. INVITE from remote host public IP to Asterisk public IP 2. 183 response from Asterisk public IP to remote host public IP 3. RTP from Asterisk public IP to the remote host private IP 4. RTP from remote host public IP to Asterisk public IP 5. RTP from Asterisk public IP to the remote host public IP Is there a way to prevent step 3 from happening? Or, is there a way to delay the invalid RTP from being sent from the Asterisk in step 3? Does anyone know why the Asterisk sends RTP to remote host private IP? I would expect NAT=YES to correct this issue. Thanks, Peter ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 ---End of Original Message- ___ 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
Re: [Asterisk-Users] NAT=YES
Mark Phillips wrote: Add canreinvite=no and reinvite=no to the relevant stanza in sip.conf Anyone that tells you to use reinvite= is confused. The option does not exist (check the source code if you don't believe me). reinvite= is one of the many Asterisk Urban Myths. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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
Re: [Asterisk-Users] NAT=YES
OK, so I have a nonexistant line in my settings. Why then when I remove it does my phone call fail? Rich Adamson wrote: FYI, there is no such thing as reinvite. Someone started using that in postings a long time ago and a few people keep posting it, but it doesn't exist. (Check /usr/src/astersik/configs/sip.conf.sample) Add canreinvite=no and reinvite=no to the relevant stanza in sip.conf Mark Klint, Peter wrote: Good morning Does anyone have experience with NAT=YES? I have the following configuration and am a bit confused as to why the Asterisk server initially sends out RTP to the remote host private IP and then switches to the public IP. Configuration Info: I have all users in SIP.CONF configured with NAT=YES Asterisk has a public IP Remote host is behind a firewall with NAT When I sniff on the Asterisk public network, I see the following. 1. INVITE from remote host public IP to Asterisk public IP 2. 183 response from Asterisk public IP to remote host public IP 3. RTP from Asterisk public IP to the remote host private IP 4. RTP from remote host public IP to Asterisk public IP 5. RTP from Asterisk public IP to the remote host public IP Is there a way to prevent step 3 from happening? Or, is there a way to delay the invalid RTP from being sent from the Asterisk in step 3? Does anyone know why the Asterisk sends RTP to remote host private IP? I would expect NAT=YES to correct this issue. Thanks, Peter ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 ---End of Original Message- ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] nat=yes
I am trying to use asterisk as a gateway between SER and the PSTN. Should the nat=yes config work with these sip.conf settings ?asterisk is trying to send it's response back to the private IP. [general] context=OUTGOING autocreatepeer=yes [Provider] type=friend username=X secret=X host=x.FakeProvider.com nat=yes --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.698 / Virus Database: 455 - Release Date: 6/2/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nat=yes in sip.con (changed subject)
Hmm. this rings a bell, try putting nat=yes in your sip.conf, I think that fixed the problem for me. (Or was the the login timed out thing? *shrug*) The manual is not very clear on what happens with nat=yes in sip.conf. Anyone here that could write a simple explanation of this option? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
