Re: [asterisk-users] SIP Ports (1000 to 2000 works)
Where is your DMZ pointed? Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Vicky wrote: Thereis definitely wrong in your setup . I have ipkall setup on my asterisk and dont have ports 1000-2000 open ( only 1-2,5060,4569 open ) . and incoming calls word fine for me . On 14/11/06, Al Bochter [EMAIL PROTECTED] wrote: No 1000 to 2000 is not a typo. Well let me put some light on this.. If you goto http://www.ipkall.com/ and your firewall is set to 1 to 2 you WILL NOT get SIP calls from http://www.ipkall.com/ DID's As soon as you OPEN ports 1000 to 2000 to the PBX Server the calls from http://www.ipkall.com/ will work fine. You DON'T have to make any changes to /etc/asterisk/rtp.conf This is what I ran into today So I guess you are right... It's a free for all on ports. Makes things harder to do. I think we need to get a better standard just to make this easier. // There's no standard - there are several different conventions adopted // by different vendors, though. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Peter Bowyer wrote: On 13/11/06, Al Bochter [EMAIL PROTECTED] wrote: Yes you are right 1-2 are rtp ports used by asterisk by default I have some that do set a custom range in /etc/asterisk/rtp.conf .. After looking around.. There were not any notes about the 1000 - 2000 port range on there website. As you know if you don't know what the ports are it no workie! And it is not good to DMZ the server. -- Now I have a handytone 386 that is set to SIP port 5060 and 5062 RTP port 5004 and 5008 You can set Random Ports to use:1024 to 65535 The handytone will work fine on the LAN But if you would moved the Handytone to the internet it would NOT work do to the firewall.. Using the asterisk defaults -- So liked I ask before"So is there any standard ports" Both sides have to be willing to negotiate a port. Maybe your handytone has its own restrictions on RTP ports? As you now know, Asterisk doesn't care as long as you specify a range in rtp.conf. 1000-2000 must be a typo as ports 1024 are reserved and privileged. There's no standard - there are several different conventions adopted by different vendors, though. http://en.wikipedia.org/wiki/Real-time_Transport_Protocol might help. Peter ___ --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 Inbound (clean). Database: 0648-1, 11/13/2006 - 11/14/2006 2:29:51 AM ___ --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] SIP Ports (1000 to 2000 works)
I was reading the posts and someone said about the default 1000 to 2000 I see in the .conf the default is 1 to 2 I found a service that gives inbound DID's in the firewall 5060 and 1 - 2 is setup no workie on the DID But when I set 5060 , 1 - 2 and (Unblocked) 1000 - 2000 Now the DID works fine. So you me what the standard is -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email ___ --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] SIP Ports (1000 to 2000 works)
actually 1-2 are rtp ports used by asterisk .. its not really compulsary .. you can set a custom range in /etc/asterisk/rtp.conf .. check ur rtp.conf what range its using and open that in firewall . Default with asterisk is 1-2 unless changed . On 14/11/06, Al Bochter [EMAIL PROTECTED] wrote: I was reading the posts and someone said about the default 1000 to 2000I see in the .conf the default is 1 to 2I found a service that gives inbound DID's in the firewall 5060 and1 - 2 is setup no workie on the DIDBut when I set 5060 , 1 - 2 and (Unblocked) 1000 - 2000Now the DID works fine.So you me what the standard is--Best regards,Al BochterBochter Services http://www.BochterServices.com/?t=EmailAre you outside of the US?Do you need to call US Toll Free Numbers?We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED](Cellular) 1-712-432-5401(Voip PBX) Free World DialUp: 780-217 EXT: 250WebSite: http://www.freeworlddialup.com/BUY and sell Coins, Silver and Goldhttp://www.bochterservices.com/?j=goldt=emailFor new and used security items http://www.bochterservices.com/?j=storet=email_securityGOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email___--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] SIP Ports (1000 to 2000 works)
Yes you are right 1-2 are rtp ports used by asterisk by default I have some that do set a custom range in /etc/asterisk/rtp.conf .. After looking around.. There were not any notes about the 1000 - 2000 port range on there website. As you know if you don't know what the ports are it no workie! And it is not good to DMZ the server. -- Now I have a handytone 386 that is set to SIP port 5060 and 5062 RTP port 5004 and 5008 You can set Random Ports to use: 1024 to 65535 The handytone will work fine on the LAN But if you would moved the Handytone to the internet it would NOT work do to the firewall.. Using the asterisk defaults -- So liked I ask before "So is there any standard ports" Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Vicky wrote: actually 1-2 are rtp ports used by asterisk .. its not really compulsary .. you can set a custom range in /etc/asterisk/rtp.conf .. check ur rtp.conf what range its using and open that in firewall . Default with asterisk is 1-2 unless changed . On 14/11/06, Al Bochter [EMAIL PROTECTED] wrote: I was reading the posts and someone said about the default 1000 to 2000 I see in the .conf the default is 1 to 2 I found a service that gives inbound DID's in the firewall 5060 and 1 - 2 is setup no workie on the DID But when I set 5060 , 1 - 2 and (Unblocked) 1000 - 2000 Now the DID works fine. So you me what the standard is -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email ___ --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 Inbound (clean). Database: 0648-1, 11/13/2006 - 11/13/2006 4:03:01 PM ___ --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] SIP Ports (1000 to 2000 works)
FRom voip-info.org# SIP on UDP port 5060. Other SIP servers may need TCP port 5060 as well iptables -A INPUT -p udp -m udp --dport 5004:5082 -j ACCEPT # IAX2- the IAX protocol iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT # IAX - most have switched to IAX v2, or ought to iptables -A INPUT -p udp -m udp --dport 5036 -j ACCEPT # RTP - the media stream iptables -A INPUT -p udp -m udp --dport 1:2 -j ACCEPT # MGCP - if you use media gateway control protocol in your configuration iptables -A INPUT -p udp -m udp --dport 2727 -j ACCEPTOpen all above ports and you should be good to go . Maybe you are recieving calls over iax and u havent opened iax2 port 4569 .. Anyway my server has all above ports opened and i have zero problems :) . On 14/11/06, Al Bochter [EMAIL PROTECTED] wrote: Yes you are right 1-2 are rtp ports used by asterisk by default I have some that do set a custom range in /etc/asterisk/rtp.conf .. After looking around.. There were not any notes about the 1000 - 2000 port range on there website. As you know if you don't know what the ports are it no workie! And it is not good to DMZ the server. -- Now I have a handytone 386 that is set to SIP port 5060 and 5062 RTP port 5004 and 5008 You can set Random Ports to use: 1024 to 65535 The handytone will work fine on the LAN But if you would moved the Handytone to the internet it would NOT work do to the firewall.. Using the asterisk defaults -- So liked I ask before So is there any standard portsBest regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Vicky wrote: actually 1-2 are rtp ports used by asterisk .. its not really compulsary .. you can set a custom range in /etc/asterisk/rtp.conf .. check ur rtp.conf what range its using and open that in firewall . Default with asterisk is 1-2 unless changed . On 14/11/06, Al Bochter [EMAIL PROTECTED] wrote: I was reading the posts and someone said about the default 1000 to 2000 I see in the .conf the default is 1 to 2 I found a service that gives inbound DID's in the firewall 5060 and 1 - 2 is setup no workie on the DID But when I set 5060 , 1 - 2 and (Unblocked) 1000 - 2000 Now the DID works fine. So you me what the standard is -- Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Goldhttp://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICEShttp://www.bochterservices.com/?j=platingt=email ___ --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 Inbound (clean). Database: 0648-1, 11/13/2006 - 11/13/2006 4:03:01 PM ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] SIP Ports (1000 to 2000 works)
On 13/11/06, Al Bochter [EMAIL PROTECTED] wrote: Yes you are right 1-2 are rtp ports used by asterisk by default I have some that do set a custom range in /etc/asterisk/rtp.conf .. After looking around.. There were not any notes about the 1000 - 2000 port range on there website. As you know if you don't know what the ports are it no workie! And it is not good to DMZ the server. -- Now I have a handytone 386 that is set to SIP port 5060 and 5062 RTP port 5004 and 5008 You can set Random Ports to use: 1024 to 65535 The handytone will work fine on the LAN But if you would moved the Handytone to the internet it would NOT work do to the firewall.. Using the asterisk defaults -- So liked I ask before So is there any standard ports Both sides have to be willing to negotiate a port. Maybe your handytone has its own restrictions on RTP ports? As you now know, Asterisk doesn't care as long as you specify a range in rtp.conf. 1000-2000 must be a typo as ports 1024 are reserved and privileged. There's no standard - there are several different conventions adopted by different vendors, though. http://en.wikipedia.org/wiki/Real-time_Transport_Protocol might help. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --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] SIP Ports (1000 to 2000 works)
No 1000 to 2000 is not a typo. Well let me put some light on this.. If you goto http://www.ipkall.com/ and your firewall is set to 1 to 2 you WILL NOT get SIP calls from http://www.ipkall.com/ DID's As soon as you OPEN ports 1000 to 2000 to the PBX Server the calls from http://www.ipkall.com/ will work fine. You DON'T have to make any changes to /etc/asterisk/rtp.conf This is what I ran into today So I guess you are right... It's a free for all on ports. Makes things harder to do. I think we need to get a better standard just to make this easier. // There's no standard - there are several different conventions adopted // by different vendors, though. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Are you outside of the US? Do you need to call US Toll Free Numbers? We can help you save money on calling US toll free numbers. Email for information: [EMAIL PROTECTED] (Cellular) 1-712-432-5401 (Voip PBX) Free World DialUp: 780-217 EXT: 250 WebSite: http://www.freeworlddialup.com/ BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Peter Bowyer wrote: On 13/11/06, Al Bochter [EMAIL PROTECTED] wrote: Yes you are right 1-2 are rtp ports used by asterisk by default I have some that do set a custom range in /etc/asterisk/rtp.conf .. After looking around.. There were not any notes about the 1000 - 2000 port range on there website. As you know if you don't know what the ports are it no workie! And it is not good to DMZ the server. -- Now I have a handytone 386 that is set to SIP port 5060 and 5062 RTP port 5004 and 5008 You can set Random Ports to use: 1024 to 65535 The handytone will work fine on the LAN But if you would moved the Handytone to the internet it would NOT work do to the firewall.. Using the asterisk defaults -- So liked I ask before So is there any standard ports Both sides have to be willing to negotiate a port. Maybe your handytone has its own restrictions on RTP ports? As you now know, Asterisk doesn't care as long as you specify a range in rtp.conf. 1000-2000 must be a typo as ports 1024 are reserved and privileged. There's no standard - there are several different conventions adopted by different vendors, though. http://en.wikipedia.org/wiki/Real-time_Transport_Protocol might help. Peter ___ --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] SIP Ports (1000 to 2000 works)
Thereis definitely wrong in your setup . I have ipkall setup on my asterisk and dont have ports 1000-2000 open ( only 1-2,5060,4569 open ) . and incoming calls word fine for me . On 14/11/06, Al Bochter [EMAIL PROTECTED] wrote: No 1000 to 2000 is not a typo.Well let me put some light on this..If you goto http://www.ipkall.com/and your firewall is set to 1 to 2 you WILL NOT get SIP calls from http://www.ipkall.com/ DID'sAs soon as you OPEN ports 1000 to 2000 to the PBX Server the calls fromhttp://www.ipkall.com/ will work fine. You DON'T have to make any changes to /etc/asterisk/rtp.confThis is what I ran into todaySo I guess you are right... It's a free for all on ports. Makes thingsharder to do.I think we need to get a better standard just to make this easier. // There's no standard - there are several different conventions adopted// by different vendors, though.Best regards,Al BochterBochter Services http://www.BochterServices.com/?t=EmailAre you outside of the US?Do you need to call US Toll Free Numbers?We can help you save money on calling US toll free numbers.Email for information: [EMAIL PROTECTED](Cellular) 1-712-432-5401(Voip PBX) Free World DialUp: 780-217 EXT: 250WebSite: http://www.freeworlddialup.com/BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=emailFor new and used security items http://www.bochterservices.com/?j=storet=email_securityGOLD PLATING SERVICEShttp://www.bochterservices.com/?j=platingt=email Peter Bowyer wrote: On 13/11/06, Al Bochter [EMAIL PROTECTED] wrote: Yes you are right 1-2 are rtp ports used by asterisk by default I have some that do set a custom range in /etc/asterisk/rtp.conf .. After looking around.. There were not any notes about the 1000 - 2000 port range on there website. As you know if you don't know what the ports are it no workie! And it is not good to DMZ the server. -- Now I have a handytone 386 that is set to SIP port 5060 and 5062 RTP port 5004 and 5008 You can set Random Ports to use:1024 to 65535 The handytone will work fine on the LAN But if you would moved the Handytone to the internet it would NOT work do to the firewall.. Using the asterisk defaults -- So liked I ask beforeSo is there any standard ports Both sides have to be willing to negotiate a port. Maybe your handytone has its own restrictions on RTP ports? As you now know, Asterisk doesn't care as long as you specify a range in rtp.conf. 1000-2000 must be a typo as ports 1024 are reserved and privileged. There's no standard - there are several different conventions adopted by different vendors, though. http://en.wikipedia.org/wiki/Real-time_Transport_Protocol might help. Peter___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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
