FYI Asterisk report shows ports in 61000 range

voip4fun

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I have some phones that are remote to my PBXIAF server. One of the phones shows registered to 5060 but the rest are in the 61000 range. Can someone please explain this? Is it some type of masquerading or caused by the remote router?
 

Hyksos

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registered from... not to.
It's NAT.
The one on 5060 and the ones on 61000 are all at the same remote location, behind the same router? that would be o_O a bit. But still functionnal.
 

Hyksos

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by that being o_O, I meant if they differ (5060 vs 61000 range) but are behind the same router... Not that them being on such different ports in 61000 is weird, that is actually necessary, not weird.
 

voip4fun

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Thank you Hyksos
They are at the same location behind the same router. I am having a problem where incoming calls to hunt group drops when answered (sip trace shows cancel from trunk side or pbx side) and wonder if this is related. I do not have these ports open in IPtables on the pbx side, I only have 5060 open. Im using Traveling Man. Should I open these ports?
 

w1ve

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Are you sending SIP OPTIONS from the phone? Your router opens a "hole" in the router for the outbound SIP Registration packet. That hole will only remain open for a while. However, if you keep sending traffic from the phone to the pbx, it keeps the hole open. If it closes during a call, the inbound packets don't know where to go, or get blocked by the router. SIP OPTIONS is essentially a PING for SIP. Most phones will call the option SIP OPTIONS, but it can also be called ping. If you don't have that option, set the registration timeout very short (like 30 seconds). You can set the qualify=true option in the Extension settings in the PBX; when you do that, Asterisk will send SIP OPTIONS packets to the phone on a regular basis. One of the other of these options will solve your problem. BTW, my PIAF is hosted, and I have 8 SIP Phones connected from the same source IP/router, and everything works nicely. BTW, 5060 is where the PBX is listening. When you are behind NAT, your phone sends a registration packet saying it is at privateIP:5060; NAT translation converts that to publicIP:somePort, where someport can be any open port number, and is usually in the high range. It's not a problem because the SIP messages will contain the proper NAT routing IPs and ports.
 

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