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New Contributor III

Re: Busy signal when dialing out - New set up

Hello,

I am having the same exact issue with TA924e connected to a flowroute trunk as well.

Except, mine works fine, but as soon as I configure ANI Substitution in the trunk with the CID i need all my FXS ports to show as, I get this error:

SIP/2.0 403 Bad au - support@flowroute.com

What i'm trying to do is override a caller ID for all FSX stations so they all show the same number. I tried all other options but nothing works. It always just shows the sip identity number as the caller id.

I need all stations to show the same phone number.

I learned that I can use a ANI Substitution to replace the caller id with the desired number in the trunk, so that any extension using that trunk to call out would show as that phone number.

But as soon as I configure it like this for example:

match $ substitute 1234567891 (i configure it in the gui)

Outbound calling stops working, I get the busy signal and that error - SIP/2.0 403 Bad au - support@flowroute.com

Any idea why it's doing it?

ANI Substitution is supposed to replace the caller ID, not the dialed number.

On the other hand, the DNIS substitution replaces the dialed number (which i don't need at this point)

Please help.

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New Contributor III

Re: Busy signal when dialing out - New set up

@acoolov

Instead of using ANI substitution, try putting an override command on your voice trunk

voice trunk T01 type sip

caller-id-override number-inbound 1002003000

Even though it says "inbound" the command will make the CPE send that DID on all outbound calls sent from that trunk

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New Contributor III

Re: Busy signal when dialing out - New set up

Thanks for your response,

I just tried that and it does not override the ext number 200 that I have put in the sip identity field. It always shows the sip identity that I put in there.

Are there any other ways to override outbound caller id?

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New Contributor III

Re: Busy signal when dialing out - New set up

acoolov

Yep, I see what you mean. I just tried it on my FXS setup and wasn't able to get it to work either right away. Guess I was thinking about the caller ID override on PRI when I wrote this earlier. Sorry about that.

I went back to the lab and got it to work in the end - I removed the caller ID override from voice trunk and added it to all of the FXS voice users instead. This puts the override DID in the SIP From and Contact fields for all outbound calls from any of the users. So try that on your side and it should work now.

voice trunk T01 type sip

no caller-id-override number-inbound

voice user 300

connect fxs 0/1

caller-id-override external-number 1002003000

voice user 301

connect fxs 0/2

caller-id-override external-number 1002003000

etc.

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New Contributor III

Re: Busy signal when dialing out - New set up

I just tried this on my TA924e

voice user 300

connect fxs 0/1

caller-id-override external-number 1002003000

But I am still getting the SIP Identity number. Here is my exact config. 1111111111 is the number I need all my lines to show when calling out. 2222222222 is the number that is entered in the sip identity for the extension.

When I call out, I still see 2222222222 showing, not 1111111111 that I want it to show.

voice user 200

  connect fxs 0/1

  password "1234"

  caller-id-override external-number 1111111111

  sip-identity 2222222222 T01 register

  sip-authentication password "1234"

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Contributor III
Contributor III

Re: Busy signal when dialing out - New set up

Normally you would have the SIP Trunk handle the registration not the FXS port.  remove the sip-identity and sip-authentication from the FXS ports leave it on the sip trunk.  Use caller-id-override to handle number change.  If your provider does not allow multiple registrations from the same identity the additional registration will be blocked or discarded.  I provided additional information in your original post.

John Wable

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New Contributor III

Re: Busy signal when dialing out - New set up

Sure, the problem is, when remove the sip identity on the fxs port it removes the trunk chosen to be used on that port. How do i choose the trunk to use for fxs port without adding sip identity? In the gui I didn't see another way to choose a trunk for the fxs port. And when i chose the trunk, i leave username and password blank so they don't authenticate, and only enter sip identity name field. But with Sipstation it things that field is username and fails. With flowroute it works.

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New Contributor III

Re: Busy signal when dialing out - New set up

You should have a "voice grouped-trunk" statement in your config that tells the CPE which trunk to route all outbound calls to.

For example:

voice grouped-trunk SIP

  description "To SIP Network"

  trunk T01

  accept $ cost 0

!

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