Unable to do attended transfer

Hi all

I have an issue that i can able to dial the call directly to the number but To the same number unable to do attendant transfer it says all circuits are busy.

I am using freepbx 16. Can anyone help me on this

How are you requesting the transfer?

Is the new leg going through a provider?

All circuits busy is a rather generic final error. You need to rather more detailed logging to find out the exact failure mode.

My initial guess is that the transferred leg is resulting in a caller ID that is unacceptable to your provider.d

Hi david55,

Below is the procedure i am following to make attendant transfer.

End user is connected to agentA , agentA enters *2 in their softphone then softphone will anounce TRANSFER and agentA enter another number to make attendant transfer.

Here we are getting all circuits are busy.

When you refer to “agentA” are these extensions in a call queue? or just normal extensions trying to transfer a call to another internal extension?

What device is the agent using (IP phone make/model, softphone app/version, mobile SIP app/version)?

Most VoIP devices have a transfer button or softkey that does a native SIP transfer, which may not have the problem you are experiencing.

Agent receives call via queue only and trying to make attendant transfer to external numbers

Hi Stewart,
Agents are using softphone

Can they transfer to an internal extension?

Yes they can tranfer to internal extensions

So if I’m understanding correctly, the user can dial out to an external number directly no problem.
They can transfer an external call from the queue to an internal extension no problem.
They CANNOT transfer an external call from the queue to an external number.

Is the at correct?

You need to look at the logs and verify the digits being sent during the transfer. This very well could be a case of the digits not matching the outbound route or not in a proper format for the provider to accept.

Yes you are correct
One more thing is they can also do blind tranfer only issue is attendant transfer

Seen the logs the numbers matching correctly.
Same number pattern working while dialing directly.

App name? Platform? Version?

For example, in MicroSIP, click || (hold), dial number to transfer to. At this point, this is a new call, Asterisk doesn’t know there will be a transfer. If this call fails, it is likely trying to use a resource that the incoming call is occupying (for example, a specific analog line). If it succeeds, select Transfer → Attended Transfer → (the incoming call) and report whether it succeeds.

The smart thing to do right now is to actually look at a verbose call log of a failed call. The “all circuits is busy” audio is played back in the macro “outisbusy” and that macro is only called when the trunk used in an outbound route doesn’t complete properly or if the trunk channels are maxed out.

The transfer is making it to the outbound routes and attempting to dial the trunk associated with said outbound routes and it is failing there. So instead of trying to guess what the issue is, looking at the actual verbose call log will give you the answers needed to figure out what is happening and why the call is hitting macro-outisbusy after attempting to dialout the trunk.

So SSH into the system, from the CLI run asterisk -rvvvvvvvvvv and reproduce the issue by answering a call from the queue and using *2 to do the attended transfer. The call should fail with “all circuits busy” recording and the console should have all the output needed which will tell you why it hit that recording and the call failed. You can copy and paste that output to something like pastebin and share the link here so we can review.

There’s the answer, CHANUNAVAIL with hangup cause 21 which is 21. Call Rejected which means one of these SIP codes was returned; 401, 403, 407, 603. Which means the call was declined/rejected by ClearRate.

Since this is a US provider, are they allowing you to send 10 digits or are they expecting 11-digits or E.164? Can you dial that number, in the same format, directly and get through?

Yes when i dial directly it is connecting.

And when i see transfer call sip flow i am getting 603 decline.

Are you using only 10 digits?

Yes same 10 digit number if i dial directly it is connecting fine

Then I would need to see more of the failed call output. This could also be a case of improper callerid being set. Does the outbound route force a callerID or just allows what ever is set during the call?