RE: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate
Thank you for your story Paul, nice work with the dialplans! I have one question, so you say that for server 2, asterisk is behind nat and you have sip clients inside and outside the nat. Which ports are you forwarding to asterisk from your firewall and in the case of sip clients outside nat, did you have to open certain ports for each client or all clients use the same? For inside clients it should be a charm! Very nice job Paul, intercity dialing and everything well connected... That was a good story.. Thx for sharing. Anton -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: Martes, 29 de Marzo de 2005 12:52 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate - Original Message - From: Anton Krall [EMAIL PROTECTED] would like to hear some actual setups and how people are solving the nat issue within scenarios like: Asterisk - nat (ports forwarded) - internet - nat - multiple voup phones I've been playing with this with my friends for awhile now. We've got four different Asterisk servers set up in four different cities: 1. 2 nics - one on internal network, other on external network. TDM400 card with 2 FXO and 1 FXS, 2 different analog lines, and a LiveVoip IAX2 dialout. Various SIP phones connected, both from within the internal network and out on the internet from behind other NATs. 2. 1 nic - behind NAT (ports forwarded). X100p with 1 analog line. Various SIP phones, internal network and from behind other NATs. 3 4. Like #2 but no X100p. All four servers are connected via IAX2 - in all cases we can dial extensions for each other's systems and the call gets dumped to the correct server. Also between server 1 2 we have local inter-city dialing working (if you dial an outside number that is local to the other city and don't put a 1 in front of the number it dumps to the other server and dials out). NAT hasn't proven to be a problem for us - the only thing we can't do as a result of all the SIP clients being natted is Reinvites - this just means that all conversation *must* go through the server as opposed to direct client-client transfer. Servers that are behind nats have the correct IP settings set in SIP.CONF. As long as I set the STUN server on the sip clients to a good working STUN server everything works like a hot damn. Nothing special regards, Paul ___ 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 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] Asterisk, SER, NAT, STUN and the whole debate
Basically, I'm forwarding the standard Asterisk ports: tcp 5060 udp 5060 udp 4569 udp 5036 tcp 5038 udp 5038 udp 1:2 I'm not sure that i needed both tcp and udp on the mgmt port 5038, but what the heck. :) In sip.conf: externip = xx.xx.xx.xx localnet=192.168.1.0 In the sip client contexts they *all* have: nat=yes canreinvite=no This is so that they can be hopped both in and out of NATs without reconfiging. No special ports being forwarded for the clients. They seem to work behind whatever NATs we throw at them without difficulties... later, Paul - Original Message - From: Anton Krall [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 5:28 AM Subject: RE: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate Thank you for your story Paul, nice work with the dialplans! I have one question, so you say that for server 2, asterisk is behind nat and you have sip clients inside and outside the nat. Which ports are you forwarding to asterisk from your firewall and in the case of sip clients outside nat, did you have to open certain ports for each client or all clients use the same? For inside clients it should be a charm! Very nice job Paul, intercity dialing and everything well connected... That was a good story.. Thx for sharing. Anton -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: Martes, 29 de Marzo de 2005 12:52 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate - Original Message - From: Anton Krall [EMAIL PROTECTED] would like to hear some actual setups and how people are solving the nat issue within scenarios like: Asterisk - nat (ports forwarded) - internet - nat - multiple voup phones I've been playing with this with my friends for awhile now. We've got four different Asterisk servers set up in four different cities: 1. 2 nics - one on internal network, other on external network. TDM400 card with 2 FXO and 1 FXS, 2 different analog lines, and a LiveVoip IAX2 dialout. Various SIP phones connected, both from within the internal network and out on the internet from behind other NATs. 2. 1 nic - behind NAT (ports forwarded). X100p with 1 analog line. Various SIP phones, internal network and from behind other NATs. 3 4. Like #2 but no X100p. All four servers are connected via IAX2 - in all cases we can dial extensions for each other's systems and the call gets dumped to the correct server. Also between server 1 2 we have local inter-city dialing working (if you dial an outside number that is local to the other city and don't put a 1 in front of the number it dumps to the other server and dials out). NAT hasn't proven to be a problem for us - the only thing we can't do as a result of all the SIP clients being natted is Reinvites - this just means that all conversation *must* go through the server as opposed to direct client-client transfer. Servers that are behind nats have the correct IP settings set in SIP.CONF. As long as I set the STUN server on the sip clients to a good working STUN server everything works like a hot damn. Nothing special regards, Paul ___ 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 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 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] Asterisk, SER, NAT, STUN and the whole debate
Any problems with RTP or voice just on one side? So as long as you use some STUN server, the RTP packets have the right IP. Did you install your own stund or are you using a public one? You didn't have to use SER at all right? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: Martes, 29 de Marzo de 2005 08:27 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate Basically, I'm forwarding the standard Asterisk ports: tcp 5060 udp 5060 udp 4569 udp 5036 tcp 5038 udp 5038 udp 1:2 I'm not sure that i needed both tcp and udp on the mgmt port 5038, but what the heck. :) In sip.conf: externip = xx.xx.xx.xx localnet=192.168.1.0 In the sip client contexts they *all* have: nat=yes canreinvite=no This is so that they can be hopped both in and out of NATs without reconfiging. No special ports being forwarded for the clients. They seem to work behind whatever NATs we throw at them without difficulties... later, Paul - Original Message - From: Anton Krall [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 5:28 AM Subject: RE: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate Thank you for your story Paul, nice work with the dialplans! I have one question, so you say that for server 2, asterisk is behind nat and you have sip clients inside and outside the nat. Which ports are you forwarding to asterisk from your firewall and in the case of sip clients outside nat, did you have to open certain ports for each client or all clients use the same? For inside clients it should be a charm! Very nice job Paul, intercity dialing and everything well connected... That was a good story.. Thx for sharing. Anton -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: Martes, 29 de Marzo de 2005 12:52 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate - Original Message - From: Anton Krall [EMAIL PROTECTED] would like to hear some actual setups and how people are solving the nat issue within scenarios like: Asterisk - nat (ports forwarded) - internet - nat - multiple voup phones I've been playing with this with my friends for awhile now. We've got four different Asterisk servers set up in four different cities: 1. 2 nics - one on internal network, other on external network. TDM400 card with 2 FXO and 1 FXS, 2 different analog lines, and a LiveVoip IAX2 dialout. Various SIP phones connected, both from within the internal network and out on the internet from behind other NATs. 2. 1 nic - behind NAT (ports forwarded). X100p with 1 analog line. Various SIP phones, internal network and from behind other NATs. 3 4. Like #2 but no X100p. All four servers are connected via IAX2 - in all cases we can dial extensions for each other's systems and the call gets dumped to the correct server. Also between server 1 2 we have local inter-city dialing working (if you dial an outside number that is local to the other city and don't put a 1 in front of the number it dumps to the other server and dials out). NAT hasn't proven to be a problem for us - the only thing we can't do as a result of all the SIP clients being natted is Reinvites - this just means that all conversation *must* go through the server as opposed to direct client-client transfer. Servers that are behind nats have the correct IP settings set in SIP.CONF. As long as I set the STUN server on the sip clients to a good working STUN server everything works like a hot damn. Nothing special regards, Paul ___ 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 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 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 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] Asterisk, SER, NAT, STUN and the whole debate
Anton Krall wrote: Any problems with RTP or voice just on one side? So as long as you use some STUN server, the RTP packets have the right IP. Did you install your own stund or are you using a public one? You didn't have to use SER at all right? Setting nat=yes does pretty much the same as a STUN server. ___ 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] Asterisk, SER, NAT, STUN and the whole debate
I tried doing a sample test with a softphone behind nat and trying to connect to my asterisk who has ports forwarded, so far, it can connect but as usual, I can hear the prompt but for example, using the echo test, I don't hear myself back. By doing a sip show peers I see the softphone connected but instead of showing using port 5060, it shows using port 64112 for example. I have nat=yes and canreinvite=no ... Any ideas? I was thinking about stun and ser but what do you guys think? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Martes, 29 de Marzo de 2005 10:28 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate Anton Krall wrote: Any problems with RTP or voice just on one side? So as long as you use some STUN server, the RTP packets have the right IP. Did you install your own stund or are you using a public one? You didn't have to use SER at all right? Setting nat=yes does pretty much the same as a STUN server. ___ 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 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] Asterisk, SER, NAT, STUN and the whole debate
Guys. Im reading a lot about ser, nat, stun, etc. And I noticed there are a lot of ways to get around nat but I would like to hear some success stories about handling nat users with multiple voip phones behind nat. I have my asterisk box behind but ports are forwarded (5060 5004 1-2 for rtp and 4569 for iax2) but still.. I can quite figure out what ser and stund have to do on this scenarios. I know ser is a sip proxy and stun helps you get your outside ip known but I would like to hear some actual setups and how people are solving the nat issue within scenarios like: Asterisk - nat (ports forwarded) - internet - nat - multiple voup phones Any good stories? Ive read the wiki, googled, etc. But I guess its time to hear what people have actually done and works. Why reinvent the wheel. ___ 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] Asterisk, SER, NAT, STUN and the whole debate
- Original Message - From: Anton Krall [EMAIL PROTECTED] would like to hear some actual setups and how people are solving the nat issue within scenarios like: Asterisk - nat (ports forwarded) - internet - nat - multiple voup phones I've been playing with this with my friends for awhile now. We've got four different Asterisk servers set up in four different cities: 1. 2 nics - one on internal network, other on external network. TDM400 card with 2 FXO and 1 FXS, 2 different analog lines, and a LiveVoip IAX2 dialout. Various SIP phones connected, both from within the internal network and out on the internet from behind other NATs. 2. 1 nic - behind NAT (ports forwarded). X100p with 1 analog line. Various SIP phones, internal network and from behind other NATs. 3 4. Like #2 but no X100p. All four servers are connected via IAX2 - in all cases we can dial extensions for each other's systems and the call gets dumped to the correct server. Also between server 1 2 we have local inter-city dialing working (if you dial an outside number that is local to the other city and don't put a 1 in front of the number it dumps to the other server and dials out). NAT hasn't proven to be a problem for us - the only thing we can't do as a result of all the SIP clients being natted is Reinvites - this just means that all conversation *must* go through the server as opposed to direct client-client transfer. Servers that are behind nats have the correct IP settings set in SIP.CONF. As long as I set the STUN server on the sip clients to a good working STUN server everything works like a hot damn. Nothing special regards, Paul ___ 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
