SOLVED: Routing DIDWW DIDs

[bcalder01]

Hi,
I am trying to route a DID from DIDWW to my server. I have another DID from SIPGate that is working properly, so access from the outside world is fine. I can route the DIDWW DID into my server to map to my SIPGate account (set up on DIDWW's control panel), but I can't get it to route directly to my server.The extension happens to be a local phone number, but the routing is working fine when I call extension to extension.

On DIDWW's site, I have set up a custom mapping of SIP/<my servers IP address> (The tech at DIDWW also had me try SIP/<my servers IP address>/<my DID>, as well as SIP/<extension number>@<my servers IP address>, but no joy). Following an article on their support site for setup with A@H & Trixbox, I have set up a bunch of trunks pointing to their servers in the US & UK with only trunk name & incoming settings defined.

I'd be fine using the SIPGate mapping, but when I create an inbound route with DID=<DIDWW DID> and CID=869504 (the number that comes up when I call the DID & it rings my IP phone), I can't get PiAF to recognize it.

I hope this isn't too inarticulate. Any logfiles/logfile snippets or other info I'm happy to provide.

Thanks in advance!

 

[jmullinix]

Try sip/<DID>@<Your IP>

 

[bcalder01]

thanks John, no go on this. I get a single ring, then silence. Looking in the logs, I don't believe the call is even getting to the server.

 

[bcalder01]

A little more ... I've been chatting with DIDWW tech support all morning (nice & responsive people) & trying different configs. Same result. I am not seeing the call making it to the server, even if I specify :5060 in the URI. When they try to call the DID, they are seeing in their logs: "[FONT=Verdana, Arial, Helvetica] SIP/<my server IP>:5060 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0)[/FONT]". They are suggesting it's a misconfig on my side, but how can it be if I'm not even seeing it get to me? By specifying port 5060, I'm sure it's passing the firewall (anyway, when I map the DID to my SIPGate account, it's routing fine), but I will parse the FW logs to be sure.

 

[jroper]

Hi

As confirmation, type sip set debug at the asterisk command prompt, you should then see lots of activity on the screen when the call is made.

This will at least confirm that the SIP messages are making to your PBX.

Joe

 

[bcalder01]

SOLVED
[FONT=Verdana, Arial, Helvetica]Our sysadmin was doing really crazy NAT stuff on the firewall, unbeknowest to me. The incoming server IP address needed to be set differently than the outgoing. I've practically pulled all my hair out over this, and it comes down to zero communication. At least it's solved.[/FONT]

 

[wardmundy]

It always comes down to networking, doesn't it. :sweatdropb:

 

[mbellot]

Try sip/<DID>@<Your IP>

John, where would you put this in FreePBX?

I tried inserting SIP/NXXNXXXXXX in the DID field for the inbound route and received two consecutive errors.

1. "This is a non-standard DID, do you want to continue?" (ok, no big deal)

2. "A slash is never allowed in the DID"

I have an older Cisco 1750 that only seems to work with the Any DID/Any CID inbound route.

Oddly enough, the call records show the call coming in on channel "SIP/NXXNXXXXXX" (followed by a "-" and a hex number (call number?)

Its an unregistered trunk, the IOS in the unit is so old it doesn't support SIP registration.

 

[jmullinix]

That is the route that I give DIDWW to route the call.

sip/<yourDID>@<yourIP>

Then I create an inbound route in FreePBX that matches <yourDID>.

 

[bcalder01]

It always comes down to networking, doesn't it. :sweatdropb:

Cripes, I think so, Ward! Thanks for all the help, everybody.