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Unread 03-17-2008, 07:22 PM   #1
wvh
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Default Multiple SipBroker Aliases

I have registered 3 sip URIs pointing to my Asterisk box. I have set up each alias to match each of my dids. However, only 1 of the 3 work. All 3 sip URIs are valid and work. The 2 that don't work just give a fast busy.

Any suggestions?
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Unread 03-17-2008, 08:43 PM   #2
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Originally Posted by wvh View Post
I have registered 3 sip URIs pointing to my Asterisk box. I have set up each alias to match each of my dids. However, only 1 of the 3 work. All 3 sip URIs are valid and work. The 2 that don't work just give a fast busy.

Any suggestions?
So if you were to call them as SIP URIs from something like Gizmo (just thinking of a software that can let you test them quickly) they ring fine....

Another way to test is to enter the SIP URI here SIPBroker - EziDial and see if you receive a call from the Echo Test App.

Are the 3 SIP URIs something like Number@yourIP, or are you using the URI of the DID providers?

It sounds like a minor mixup that just needs to be tested.....

On a sidenote:
One suggestion I have which may probably prove to be a lot more work (and probably beyond the issues here), but that is quite neat to play with, is if you have your own domain, you may be able to adjust the SRV records to point to your asterisk server and get your own SipBroker SipCode assigned to your domain
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Unread 03-17-2008, 11:02 PM   #3
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Quote:
Originally Posted by emoci View Post
So if you were to call them as SIP URIs from something like Gizmo (just thinking of a software that can let you test them quickly) they ring fine....

Another way to test is to enter the SIP URI here SIPBroker - EziDial and see if you receive a call from the Echo Test App.

Are the 3 SIP URIs something like Number@yourIP, or are you using the URI of the DID providers?

It sounds like a minor mixup that just needs to be tested.....
All three sip URIs work fine using the echo test. They also work via Enum lookup. (However, the two don't work via SipBroker ENUM server *013 or sipbroker *011). They are name@myasteriskurl. I then have an incoming route for each name. All three are set up the same. That is why I can not figure out what is wrong. I connected to the asterisk CLI and I do not see anything coming in when I test the two aliases that are not working via SipBroker.

Last edited by wvh; 03-17-2008 at 11:28 PM.
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Unread 03-17-2008, 11:25 PM   #4
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All three sip URIs work fine using the echo test. They also work via Enum. They are name@myasteriskurl. I then have an incoming route for each name. All three are set up the same. That is why I can not figure out what is wrong. I connected to the asterisk CLI and I do not see anything coming in when I test the two aliases that are not working via SipBroker.
Why the aliases are not working puzzles me too then....can you think of any significant difference between the one line that works and the other two that do not ??

On a sidenote, if the numbers are already setup for ENum, you do know you can just call any SipBroker access number, and then dial the number (which has been setup for ENum) directly in CountryCode-Number format and have it ring, without having to go through the trouble of getting an alias

Make sure the ENum record is showing up right before trying the above SIPBroker - ENUM Lookup
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Unread 03-17-2008, 11:31 PM   #5
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They look right via the enum lookup. The enum look up also works right on my asterisk box.
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Unread 03-17-2008, 11:44 PM   #6
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They look right via the enum lookup. The enum look up also works right on my asterisk box.
Ok, so try calling them from SipBroker, dial an access number, then when prompted enter one of your numbers as they are registered for ENum.... (the trick is to enter the full CountryCode-Number)

Does that work....

And most importantly does that serve your purpose instead of the alias?

If it helps for testing, you can also use this to dial via SipBroker Free VoIP Calls
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Unread 03-18-2008, 12:23 AM   #7
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No, that does not work either. If I call via an access number, then use *0131xxxxxxxxxx or *0111xxxxxxxxxx, only one of my three numbers work.

However, if I call from my asterisk box, I see a successful enum lookup and the call completes via enum for all three.
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Unread 03-18-2008, 06:35 PM   #8
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Thanks for the help in figuring this out. I finally got it working in my Asterisk Freepbx setup. I was using inbound routes to point multiple dids and aliases to one line on my ata. The ENum worked, but Sip Broker did not with this configuration. To get this to work, I added a custom extension for each alias. I then routed each custom extension to my extension for my ata.
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