Multiple Registration of SIP Trunk Lines

FreePBX Providers
1

Hi All,

I have an SIP trunk and I have performed the configuration;

  1. Establish the SIP Trunk (using the pilot number - 084xxx904)
  2. Configure the Outbound route (using the trunk above. calls are going through)
  3. Configure inbound route (calls to 084xxx904 was able to hit IVR )

Here is the problem, there are other lines 084xxx905 - 934. They are configured in the inbound route but non of them could receive calls.

Provider said the lines (905 - 934) had to be registered before I can receive calls. Can anyone help me with the config to achieve this?

Thank you

2

Do these calls hit the PBX at all?

3

No. It did not except the 904

4

Would appear to be your provider’s routing, what sip channel driver are you using and do the calls appear in sngrep? (sngrep sees traffic at the kernel level before your firewall or forwarding)

5

They seem to be providing a service for direct use by phones, rather than by a PABX.

Given that you should be using chan_pjsip on a new system, one way of doing this would be add multiple type=registration sections, referencing the same endpoint, as pjsip.conf customisations.

6

I am using Chan_Sip.

My guess is as good as yours.

I debugged asterisk and I did not see the call at all.

7

@david55

Can you shed more light on the pjsip.conf.

I appreciate

8

if asterisk never sees the call then I suggest your problem lies elsewhere , you need to fix that first.

9

The provider has, effectively, said that they will not present calls unless there has been a registration for the specific DID, not just for the pilot number. That is a plausible explanation of why there no call to be seen.

10

Given that there is a lot of information already available on the use of chan_pjsip with FreePBX, as it is the preferred SIP channel driver, I’m not sure why I should repeat that. Also, I’m a raw Asterisk person, and I might overlook ways of doing things with the GUI.

Note that I said customisations, and customisations never go directly in the default Asterisk configuration file, when using FreePBX.

For chan_sip, it is done with register lines in the general section, but, as I pointed out, you should not be using chan_sip for new installations.

11

That is the main focus of my post.

Is there any approach to achieving this? Is there a way to register multiple sip trunk lines?

12

If you are not prepared to customise the channel driver configuration files, I think you could probably define each number as a separate trunk, in the GUI. You would probably find that incoming calls are only matched against one of them, so the channel name would not reflect the exact number used.

13

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