1. Change The Cid Of Extension Sip Freepbx Number

Whichever it is we create the extension in FreePBX the same way. 1 – Log in to FreePBX and select Applications / Extensions from the menu bar at the top. 2 – Ensure “Generic CHAN SIP Device” is selected and click submit. 3 – Enter extension details. These are the details you will use in you physical VOIP extension or softphone. Navigate to Applications - Extensions and on that page click Add New Extension - Add New ChanSIP Extension. Give your extension an extension number in the User Extension field; additionally, you have the option to put an extension-specific caller id name in the Display Name field and caller ID number in the Outbound CID field. You will need to create an extension for your ATA so it can register to FreePBX and receive/make calls on behalf of your fax machine. Navigate to Applications - Extensions and on that page click Add New Extension - Add New ChanSIP Extension.

Fallout 4 vr mods not working. I believe this is a function of how your SIP provider authenticates the trunk. I've seen SIP providers authenticate the trunk based on the public ip of your PBX, based on a register string, or in some cases based on your main DID in addition to one of these. As you say, if you change the outbound CID for an extension and make an outbound call, the CID you enter shows when you make an outbound call to an external number. In that case, the SIP provider knows the call originated from your PBX and allows it.

But, when a call comes in from the outside, hits your PBX, and then is pushed out to a cell phone, the call did not originate from your PBX in this case. Many SIP providers I have seen only allow that call to be pushed back out via Follow Me if you use a CID that is part of your SIP trunk / will not honor a CID change for the outbound call in this case. One company I know couldn't get follow me to work at all without using the main DID for CID of Follow Me calls to cell phones because the SIP provider would reject the calls.

Does Asterisk / FreePBX support the ability to pass the caller ID of an inbound caller to a remote support agent (on a cell phone)?

Our work has a queue for incoming calls which contains 'remote agents' (people on cell phones). To the cell phone agents, all calls appear to be coming from our main number (385-111-1111). We would like the calls to appear to be coming from the caller (201-555-5555).

This is not a problem with our SIP trunk provider. In the past we used different PBX software, with the same SIP trunk provider, and it was able to set the Caller ID properly. Extensions are capable of setting and passing arbitrary Caller ID, only calls from queues retain the main number.

Outgoing PEER Details:

I've manipulated so many settings that I've come to wonder if Asterisk / FreePBX simply does not support this. Has anyone successfully been able to do this?

Change The Cid Of Extension Sip Freepbx Number

2 Answers

Asterisk certainly does. Capture the CID in a dialplan variable at the beginning of the call and set the outbound CID to the same value before passing it on.

There's no direct way to do this within the FreePBX GUI but there is a workaround:

1. Set up a virtual extension
2. Enable follow-me on the extension, add the mobile number to the follow-me list
3. Set the follow-me CID mode to default
4. Ensure the queue's agent restrictions allow the use of follow-me numbers
5. Have the agent log into the queue using the virtual extension instead of their mobile number

The default behaviour for the follow-me extension is to pass the incoming caller ID out. So, some flexibility is lost (mobile numbers have to be changed in follow-me settings) but it does allow the desired behaviour.

Asterisk supports setting the callerid for all outgoing or redirected calls. I did this with v1.8 and v13.7 as I'm facing the exact same requirements.

This feature depends on the provider and the contract they setup with you. My Provider calls it 'Special Arrangement / Clip no screening'. In my case they use 'P-Asserted-Identity' to find callerid.

I had to set the following options in the outgoing sip trunk in sip.conf:

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