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Unread 06-25-2009, 03:30 PM   #1
javiercm
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Arrow Probems with all Servers (us, eu, au)

Hello, I always use the us server. The last week I could make and receive calls, but now I can't receive calls. If I make a call I hear the other person but the other person can hear me.


So i tried to change the server to eu.voxalot.com and now all work fine but now I have a new problem. If I use eu server I get FUP exceeded.


So i have just to try changing the server to au.voxalot.com and now I get the same problem that using the us server.

Help me, please !!!

Sorry for my English. ;-)
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Unread 06-25-2009, 07:17 PM   #2
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Eventually Betamax may add all proxies to to their white list. Until then you may be able to avoid the FUP problem by choosing a specific proxy. The five I know of are

proxy01.us1.voxalot.com
proxy02.us1.voxalot.com
proxy01.eu1.voxalot.com
proxy02.eu1.voxalot.com
proxy01.au1.voxalot.com


I can't say for sure why you can't receive calls, however it could be due to an issue with your local router, perhaps the NAT session (for the SIP packets) has expired. What kind of SIP phone are you using?

You can do a simple test; after configuring for one of the above proxies restart your SIP phone and within 3 minutes try to receive a call. If you can receive a call within the first 3 to 5 minutes, but not thereafter, then a NAT session expiry problem is indicated.
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Unread 06-25-2009, 11:28 PM   #3
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thank you boatman, you are the best !!!!


I have change the proxy server and ...

proxy01.eu1.voxalot.com => I can do free calls and receive calls.
proxy02.eu1.voxalot.com => I can't hear the other person and FUP exceeded.


So now I 'm using proxy01.eu1.voxalot.com, but I have some questions.
Why have I so many problems?

I have a Linksys router (wrt54g) and two ATAs Linksys PAP2T NA ...


INTERNET
||
CABLE-MODEM
||
ROUTER WRT54G (192.168.2.1)
||
||
||<===> PC
||
||
||<===> PAP2T (1) ====> Line 1 and Line 2
||
||
||<===> PC
||
||
||<===> PAP2T (2) ====> Line 3 and Line 4


______ PAP2T (1) ______

IP: 192.168.2.51
RTP PORT MIN : 16384
RTP PORT MAX: 16482
LINE 1 PROXY PORT:5060
LINE 2 PROXY PORT:5061


______ PAP2T (2) ______

IP: 192.168.2.201
RTP PORT MIN : 16484
RTP PORT MAX: 16582
LINE 3 PROXY PORT:5062
LINE 4 PROXY PORT:5063


______ ROUTER ______

PORT RANGE FORWARD 16384-16482 ===> 192.168.2.51
PORT RANGE FORWARD 5060-5061 =====> 192.168.2.51
PORT RANGE FORWARD 16484-16582 ===> 192.168.2.201
PORT RANGE FORWARD 5062-5063 =====> 192.168.2.201



Is my configuration OK? some times all work fine, but no always....
us servers don´t work for me.

Thank you.

Last edited by javiercm; 06-25-2009 at 11:31 PM.
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Unread 06-26-2009, 02:39 AM   #4
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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.
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Unread 06-26-2009, 08:22 AM   #5
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-Where are you?
I'm in Castellón (Spain-Europe)

-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?

I only want to use a server who works fine for me. When a server don't work sometimes only can receive calls, sometimes only can make calls, sometimes only one perrson can hear the other person. This situation is random.

-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.

Yes I want say LINE 4 SIP Port:5063

-"Proxy" refers to Voxalot's end which is always 5060 or one of a few supported alternate SIP ports.

I don't understand this. If the end port is always 5060 then I have routed ports incorrectly ....

ATA2 Line 4: 5063 <----> 5062 ROUTER 5063 <----\/\/\/\---->5060 voxalot

I thought that this work of this way...

ATA1 Line 1: 5060 <----> 5062 ROUTER 5062 <------------->5060 voxalot
ATA1 Line 2: 5061 <----> 5062 ROUTER 5062 <------------->5061 voxalot
ATA2 Line 3: 5062 <----> 5062 ROUTER 5062 <------------->5062 voxalot
ATA2 Line 4: 5063 <----> 5062 ROUTER 5062 <------------->5063 voxalot

thank you for your help and your time

Last edited by javiercm; 06-26-2009 at 08:24 AM.
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Unread 06-26-2009, 12:40 PM   #6
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Every packet has a "destination port number" and a "source port number". Ordinarily port 5060 is used for SIP packets to/from Voxalot SIP servers. Therefore if you want to use proxy01.eu1.voxalot.com you would set

Proxy: proxy01.eu1.voxalot.com:5060
Outbound Proxy: proxy01.eu1.voxalot.com:5060

Because 5060 is the default SIP port so you can also use
Proxy: proxy01.eu1.voxalot.com
Outbound Proxy: proxy01.eu1.voxalot.com

Also set
Use Outbound Proxy: yes
Use OB Proxy In Dialog: yes

If you are using DNS SRV then you can't specify the port number because will disable the ATA's DNS SRV feature.

Once the Betamax FUP and other issues have been resolved you may want to use DNS SRV for better reliability.

You don't need to forward any SIP ports in your router unless you wish to avoid using a STUN server.
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Unread 06-29-2009, 08:22 AM   #7
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thank you boatman,


The last two days I'm testing the calls as you said me.

__________________________________________________ ____
5060 is the default SIP port so you can also use
Proxy: proxy01.eu1.voxalot.com
Outbound Proxy: proxy01.eu1.voxalot.com

Also set
Use Outbound Proxy: yes
Use OB Proxy In Dialog: yes
__________________________________________________ ____


and now all work fine, only fail the calls from voxalot to voxalot, I have this dail plan _ZXXXXX *010${EXTEN} SIP Broker, some times it works, but other I cant hear the other person or vice versa. I also have test dial *010XXXXXX directly in my phone and it also fail.


Thank you again.
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Unread 06-29-2009, 04:47 PM   #8
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Quote:
Originally Posted by javiercm View Post
sometimes it works, but other I cant hear the other person or vice versa.
It's a mystery to me. I can only suggest changing to a different stun server, or having someone check the SIP SDP packet sent by each of the ATAs involved in this problem. If you want me to check that, PM me the Voxalot numbers to call. To check the SDP packet someone just has to pick up the phone when called, however they don't actually need to speak.
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