Hello, we have an external PBX connected to FreePBX by SIP trunk. We would like to be able to send a call directly to the external PBX, and then have them able to hook flash (or some feature code) to get dial tone so they can send the call to an outside number.
No problem doing this with extensions, but is there a way to do this on a trunk?
Your request is pretty vague. Please give a detailed example of the desired call flow.
Where is the original caller (on a FreePBX extension, calling into FreePBX from the outside, etc.)?
Who is the original caller (a customer, an authorized employee accessing DISA, etc.)?
Is there a call transfer involved e.g. A gets connected to B, who then connects A to C?
Who is “them”?
One hosted FreePBX with a premise based NEC PBX attached using a SIP trunk. There are several remote worker extensions attached to this FreePBX.
A call comes to the NEC over the FreePBX SIP trunk. The party that answered the call wants to transfer the call to an outside DID. Is there a way? Is my explanation confusing?
It still is a little. Please correct my understanding if wrong:
An external caller dials a DID that is on your FreePBX. Based on the number called or an IVR option selected by the caller, FreePBX routes the call to the NEC and it rings a local extension. A person answers the NEC extension and now wants to transfer the caller to an external number.
Assuming the above, there are two options:
One is you do nothing special and use the normal call transfer facility of the NEC, e.g. Transfer button on the phone. This will make a new outgoing call and the NEC will bridge the calls together. Depending on the new number dialed and how the NEC is configured, it may route the outgoing call via your FreePBX.
Or, you can add ‘t’ to the Asterisk Trunk Dial Options for the trunk to the NEC. The NEC user can dial *2 and hear Asterisk dial tone, dial the transferred-to party, speak with him if needed, then hang up and Asterisk will bridge the two external parties together.
These options also differ on what gets logged on each system, how calls can be recorded, etc.
A bit more. As it turns out the connected PBX is a Panasonic, not an NEC. The handsets do have a transfer feature.
I do have the “t” in the trunk setup.
The transfer function uses two “trunks” connection/Panasonic trunk licenses. The goal here is to figure a way to compete with https://info.panasonic.com/nsvconnect/
Their SIP trunks allow a user to press *6, get outside dial tone and make an unattended transfer of the call which removes it from the Panasonic PBX licensed trunks.
So what goes wrong when you press *2 on the Panasonic handset? If nothing happens and nothing gets logged on Asterisk, check the Panny for settings related to Outbound DTMF on inbound calls, and DTMF transmission mode on the trunk.
Otherwise, post a log of a failing call.
Thank you. I’ll get to try this tomorrow and report back
I’m a bit embarrassed. I never thought to test this first on the trunks. The *2 feature works out of the box on trunks too. I assumed (I strive to never use that word) that trunks would be an issue. I went looking for a problem that did not exist…
Thank you again for your advice and help!
Err, less embarrassed. We have two FreePBX instances with two different customers.
This works on one FreePBX and not on another. I’ll test the one not working and report back.
Check your feature codes… it could be disabled in there. It could also be disabled/enabled in the transfer options on the trunks/extensions.
Hi, Still working on this. The feature code is on and correct. We are using a trunk in this instance.
What/where is the transfer option in trunks?
Where is the transfer option in extensions?
I’m not familiar with these at all.
The trunk is set "Asterisk Trunk Dial Options “t” and set to System. Is there more I can do?
I don’t know what might be wrong. Check that the Panasonic PBX is sending DTMF in the same format that Asterisk is expecting.
Since you have a system that does work, make test calls from each and compare the logs. For this test, turn on DTMF logging in Asterisk Logfile Settings.
This works out of the box! The customer site at issue still cannot seem to test this out. That is why I wrote in the first place. We have 3 other of their pbxs connected to the same instance and they all work fine.
Thank you all for weighing in to advise me
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