European Server
Gegroet,
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Great! Just some questions. (I'm not such a big expert in SIP). 1/ Does logging in on this server instead of the US server affect the SIP URI. Is it still "SIP:number@voxalot.com" or has it become "SIP:number@eu.voxalot.com"? 2/ As you say that the server is designed to be "fully operational" for SIP-calls between voxalot-users; why are there different SIP-broker id for voxalot-servers in the US, AU and EU? Can you explain the difference in set-up between these two kinds of kinds of connections (inside the voxalot domain and when using SIPbroker)? Many thanks in advance, Cheerio! Kr. Bonne. |
Good ping in Norway, echo test is much more responive now :D
Pinging eu.voxalot.com time=39ms Pinging voxalot.com time=117ms Those of you complaining about ping time should post your location as well... :rolleyes: |
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Gegroet,
First of all, thank you for replying. I'm still learning about how SIP really works so please excuse me additional questions. Quote:
What exactly do you mean with "a more direct path"? I did some tests (I like to find out things myself before annoying other people with questions :-)) where I use the SIP-client on my laptop and run "tcpdump" at the same moment to verify what really happens. The I made a PSTN-call to the SIP dail-in number here in Belgium (packetnet.be) and made a call to *010<my-number> and *031<my-number> In both cases, I got the same thing: - a SIP-message from eu.voxalot.com telling me that there is an incoming call - an outgoing UDP-flow from my client to packet.be for the voice-call. Now, of course, I cannot see what happened earlier. I don't know what happened in the earlier stages when I made the call to *010... But can I assume that the difference is only in the "control" part (i.e. the exchange of packets to do the call-setup); but that for the call itself it does not make any difference at all. Is the situation any different if I would configure my client to use "strickt outbound proxy"? Again, I'm sorry for the question. I'm still trying to really understand how SIP works "under the hood". Cheerio! Kr. Bonne. |
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It seems to work differently in my case. I did a tcpdump during both inbound (PSTN to SIP) and outbound (SIP to PSTN) calls and the RTP flow seemed to go through "hosted-by.leaseweb.com" (which is the European Voxalot server) instead of my Italian SIP provider (Messagenet.it) servers.
These were my settings at the time of the experiment: - my internet provider is Fastweb.it, which puts all its users behind a NAT; - my SIP client is registered with Voxalot only; STUN is set to stun.voxalot.com.au; - Messagenet.it, my Italian SIP provider is entered in Voxalot's "Providers" list and both "Active" and "Register" are set to "Yes"; - my Voxalot dial plans point to Messagenet for the relevant outgoing calls; - I dialled PSTN numbers from my SIP client to test outbound calls and dialled the PSTN number associated with my Messagenet account from a landline and answered them with my SIP client to test inbound calls. In both cases UDP traffic seemed to go through Voxalot. I would like to have my voice traffic go through the relevant SIP proider instead, both in order to avoid burdening Voxalot's servers and to reduce latency. Can you help? Thank you, Carlo |
Hi Martin, today I have migrated all my providers from us.voxalot.com to eu.voxalot.com and I have changed proxy address in my linksys ATA but some registered numbers doesn't ring.
Please can you verify if all my providers are correctly migrated on eu.voxalot.com ? My userid is 902830. Thanks and continue like this - very good service. Riccio Vittorio |
Hello Riccio,
Your providers and migrated settings look very good. My only thought is that allow an hour or so for your providers to remove the old settings from their cache. |
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I think we will need to set aside some time to reproduce your test in our test environment. I have added this as an item in our bugzilla database. |
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