Voxalot / SIP Broker Support Forums - View Single Post

boatman · 06-26-2009, 02:39 AM

Where are you? Would you prefer to use one of the US servers if possible? When you say the US servers don't work for you, do you mean to receive calls, make outbound calls, or both?

When you say (for example) "LINE 4 PROXY PORT:5063" I think you mean "LINE 4 SIP Port:5063". If so then your settings look OK. "Proxy" refers to Voxalot's end which is always 5060 or one of a few supported alternate SIP ports.

When one is testing receiving calls, and has recently registered to one of Voxalot's proxies (proxy A), but then switched to a different proxy (proxy B), it's possible for the calls to be received through proxy A. This is because you will be registered at both proxies for a certain time, and NAT route is still open to both. Both proxies send the call to your ATA. Your ATA takes the call from whichever proxy's packet arrives first and rejects the second call with "BUSY HERE". I don't have time to test just now, but I think this situation could even cause the inbound call to fail completely. In any case, to avoid complications when testing inbound calls through various proxies you should change your SIP port in your ATA each time you change proxies. As you may know, you get to choose the port number for your end, you can choose almost any port number. Again, changing the port number is only helpful when switching quickly between proxies and testing inbound calls.

Of course frequently changing the SIP port on your end can be a hassle if you also have to change port forwarding in your router. Therefore I would suggest you turn off all port forwarding related to your ATAs and use the following settings, at least during the time you are testing call reception through different proxies.

set the following:

(under SIP tab)
Handle_VIA_received: yes
Handle_VIA_rport: yes
Insert_VIA_received: yes
Insert_VIA_rport: yes
Substitute_VIA_Addr: yes
Send_Resp_To_Src_Port: yes
STUN_Enable: yes
STUN_Test_Enable: yes
STUN_Server: stun.voxalot.com:3478 (or any STUN server such as 'stun01.sipphone.com:3478' or 'stun.sipgate.net:10000')
NAT_Keep_Alive_Intvl: 179

(under Line_1 and Line_2 (or PSTN_Line) tabs)
NAT_Mapping_Enable: yes
NAT_Keep_Alive_Enable: yes
NAT_Keep_Alive_Msg: (this setting should be blank)
NAT_Keep_Alive_Dest: $PROXY
Register Expires: 3600

If you are trying to use the "Symmetric NAT handling" option in your Voxalot account to make Voxalot carry your voice packets (not a good idea unless it's really needed) then make the following changes to the above settings.

(under SIP tab)
STUN Enable: no

(under Line_1 and Line_2 (or PSTN_Line) tabs)
NAT Mapping Enable: no

06-26-2009, 02:39 AM	#4
boatman Senior Member Join Date: Jul 2007 Location: Oregon, USA Posts: 365 Thanks: 17 Thanked 77 Times in 64 Posts	Where are you? Would you prefer to use one of the US servers if possible? When you say the US servers don't work for you, do you mean to receive calls, make outbound calls, or both? When you say (for example) "LINE 4 PROXY PORT:5063" I think you mean "LINE 4 SIP Port:5063". If so then your settings look OK. "Proxy" refers to Voxalot's end which is always 5060 or one of a few supported alternate SIP ports. When one is testing receiving calls, and has recently registered to one of Voxalot's proxies (proxy A), but then switched to a different proxy (proxy B), it's possible for the calls to be received through proxy A. This is because you will be registered at both proxies for a certain time, and NAT route is still open to both. Both proxies send the call to your ATA. Your ATA takes the call from whichever proxy's packet arrives first and rejects the second call with "BUSY HERE". I don't have time to test just now, but I think this situation could even cause the inbound call to fail completely. In any case, to avoid complications when testing inbound calls through various proxies you should change your SIP port in your ATA each time you change proxies. As you may know, you get to choose the port number for your end, you can choose almost any port number. Again, changing the port number is only helpful when switching quickly between proxies and testing inbound calls. Of course frequently changing the SIP port on your end can be a hassle if you also have to change port forwarding in your router. Therefore I would suggest you turn off all port forwarding related to your ATAs and use the following settings, at least during the time you are testing call reception through different proxies. set the following: (under SIP tab) Handle_VIA_received: yes Handle_VIA_rport: yes Insert_VIA_received: yes Insert_VIA_rport: yes Substitute_VIA_Addr: yes Send_Resp_To_Src_Port: yes STUN_Enable: yes STUN_Test_Enable: yes STUN_Server: stun.voxalot.com:3478 (or any STUN server such as 'stun01.sipphone.com:3478' or 'stun.sipgate.net:10000') NAT_Keep_Alive_Intvl: 179 (under Line_1 and Line_2 (or PSTN_Line) tabs) NAT_Mapping_Enable: yes NAT_Keep_Alive_Enable: yes NAT_Keep_Alive_Msg: (this setting should be blank) NAT_Keep_Alive_Dest: $PROXY Register Expires: 3600 If you are trying to use the "Symmetric NAT handling" option in your Voxalot account to make Voxalot carry your voice packets (not a good idea unless it's really needed) then make the following changes to the above settings. (under SIP tab) STUN Enable: no (under Line_1 and Line_2 (or PSTN_Line) tabs) NAT Mapping Enable: no Last edited by boatman; 06-26-2009 at 02:47 AM.