brent clements
2005-Aug-02 04:03 UTC
[Asterisk-Users] This should work right??? Any caveats that you guys know about?
Hello, long time lurker, first time writer.... We have the following set up ITSP | | Internet | | Cisco 2600 | | Switch----Asterisk Server running 1.0.9(has public ip) | | Cisco 515e Pix Firewall running Pix OS 5.3(run's a class c 1-to-1 nat and pat) | | Grandstream GXP-2000(run latest fw from grandstream site 1.0.1.9) The grandstream registers with the public asterisk server fine. I even see one of the dynamic nat addresses being assigned. The Pix Firewall has sip fixed up and all VOIP related ports are wide open. This is the issue: We can make outgoing calls, but we can't receive calls when the grandstream is behind the firewall If we move the grandstream in front of the pix and give it a public ip, everything works fine. What is even wierder is the fact that one of our network users who is behind the pix firewall can use ATT's VOIP service just fine. Are there any things I should be looking for? In general is the setup above pretty common? I've looked through the Wiki and searched google many times but nothing that can give me any pointers. Thanks!
hi, I am going to open up a call center starting with 5 and expanding to 20 seats in 3 months. I have decided to use asterisk. I don't think I need FXO or any other card from digium. If you have any document regarding setting up a call center with asterisk then please let me know. What additional things I need to buy except the server (pentium 4 with 1gb ram). thanks in advance, Zeeshan.
Ashish Raikwar
2005-Aug-02 06:07 UTC
[Asterisk-Users] This should work right??? Any caveats that youguys know about?
hi Solution of your problem is in this article which i am pasting from an online document.... A SIP phone usually registers with a SIP proxy. This message comes from the inside of the NAT to the server on the outside. Now, there's an open connection in the NAT device. As soon as there's no more packets on that connection, the NAT device cancels the connection and forgets all about it. The trick is to keep the packets flowing, forcing the NAT device to keep the connection open. Some phones send NAT "keep-alive" packets by themselves. X-lite and Sipura have this feature. If the phone can't do it, configure Asterisk to do it. Setting "qualify=yes" in the [peer] section for this device, Asterisk starts sending packets to the device, keeping the NAT connection open. You will also be able to see some timing for packets between Asterisk and the phone when you do "sip show peers" at the CLI. Now, when Asterisk wants to place a call to the phone, the NAT welcomes the packets and forwards them happily to your phone. Conclusion: If Asterisk is on a public IP address and your phone is on the inside of a NAT device, we need to keep the NAT connection open by frequently sending dummy packets between the devices. This will keep the connection open for incoming calls. ----- Original Message ----- From: "brent clements" <bcasterisktechlist@gmail.com> To: <Asterisk-Users@lists.digium.com> Sent: Tuesday, August 02, 2005 4:03 AM Subject: [Asterisk-Users] This should work right??? Any caveats that youguys know about? Hello, long time lurker, first time writer.... We have the following set up ITSP | | Internet | | Cisco 2600 | | Switch----Asterisk Server running 1.0.9(has public ip) | | Cisco 515e Pix Firewall running Pix OS 5.3(run's a class c 1-to-1 nat and pat) | | Grandstream GXP-2000(run latest fw from grandstream site 1.0.1.9) The grandstream registers with the public asterisk server fine. I even see one of the dynamic nat addresses being assigned. The Pix Firewall has sip fixed up and all VOIP related ports are wide open. This is the issue: We can make outgoing calls, but we can't receive calls when the grandstream is behind the firewall If we move the grandstream in front of the pix and give it a public ip, everything works fine. What is even wierder is the fact that one of our network users who is behind the pix firewall can use ATT's VOIP service just fine. Are there any things I should be looking for? In general is the setup above pretty common? I've looked through the Wiki and searched google many times but nothing that can give me any pointers. Thanks! _______________________________________________ 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