I’m probably doing this wrong… Call flow proceeds through an incoming IVR Into first ring group, if that isn’t answered it goes to a second ring group and both of those use call confirm to positively accept the calls. All of that works perfectly fine.
The last step however is an “overflow” to an external call handling agency. So the requirements are;
Announce to caller that the final destination is being tried
Announce to destination the nature of the call
DO NOT require the destination to confirm the call. E.g. the intent is to connect the caller immediately after the call handling agent has heard the announcement telling them where the call comes from.
So I tried defining a third ring group with only the call handling extension as a destination. “Confirm Calls” is switched OFF in both the call handling extension and in the Ring Group.
The caller is getting the correct announcement and then Music on Hold as intended.
The far end is hearing the Remote Announcement but the call handler then needs to confirm the call by pressing 1.
So it’s acting as if the “Confirm Calls” from the earlier Ring Group in the flow is still in effect, permitting the remote announce from RG#3 to be heard. But I don’t want to confirm this last leg. And if I somehow switched off call confirmation completely I suspect the remote announcement would also be removed.
So - if I want to play a remote announcement before transferring a call to an extension without call confirmation, what strategy could/should I adopt?
Would simply setting additional options in the “Advanced” tab do the trick? This thread suggested that the Dial application would allow this, so perhaps setting the call handling extension’s Asterisk Dial Options to “HhTtrA(custom/my_remote_accouncement)” would do the trick?
I’m guessing that setting the dial options for the extension will not propagate through follow-me or call forwarding. If you have that trouble, you could configure the A(custom/my_remote_announcement) on a trunk (that is otherwise a copy of the regular trunk) and route the call there via e.g. a special prefix.
However, why is this announcement needed? Normally, when a call is forwarded to an answering service, their system captures the number from which the call was forwarded (as well as the number of the original caller). Depending on the sophistication of the service, their phone will display your company number, your company name, or will automatically pop up a CRM screen for your company.
This scheme is typically known as RDNIS; see https://www.voip-info.org/rdnis/ . With a SIP trunk (and most providers), you put your company number in the Diversion header. If you are lucky, simply turning on Generate Diversion Headers in Advanced settings will magically make it work.
Hi @Stewart1, thanks for the swift reply. The reason is that the boss wants it but mostly to audibly prepare the call handler as they deal with other customers. I think I may have found (a/the) solution however - set up a Queue.
It seems that a Queue can play a “Join Announcement” to the caller (Goal 1), then I can add an external number directly as a Static Agent, and in Timing and Agent Options I set an “Agent Announcement” (Goal 2) and provided I leave “Call Confirm” set to No in the General Settings, it doesn’t seem to require input from the agent, which achieves Goal 3.
So I’m off to try putting this into the call flow to try it out for real!
I hadn’t thought of the queue, which is much cleaner than the extra trunk scheme. However, if you don’t want RDNIS or can’t make it work with your provider, make sure that the external service is prepared for what you are doing. I suspect that the great majority of their customers either use ‘call forward on no answer’ or turn on call forwarding outside of business hours; in either case they don’t hear an announcement and rely on seeing the forwarding number.
I will certainly ask about the RDNIS setup, but I believed that our SIP/PSTN gateway provider (Voipfone in the UK) doesn’t support us setting dynamic outgoing CallerID of other options when making outbound calls to the PSTN (which makes sense as it’s easily abused). We’re not sending SIP calls to the destination call centre you see. As a result we have for some time simply passed on a single CallerID to the call centre agents, and that hasn’t changed - this is just an additional prompt to get them in the right frame of reference for the calls.
If you don’t need to pass the original caller’s number to the call centre, you don’t really need RDNIS. They should recognize your company’s number in the caller ID and answer appropriately.
However, if you forward calls to user mobiles, passing the number of the original caller is quite useful. Check with Voipfone if they permit this. Some providers do but require special header setup.
You might consider a supplementary provider that doesn’t restrict the caller ID. Take a look at https://www.voxbeam.com/ . You get a small credit at signup, so you can test without making a payment or providing any financial info. Some typical rates (Platinum route in US$; £ accounts are of course also available):
4420 GB United Kingdom Fixed - London 0.0022
44 GB United Kingdom Fixed 0.0047
447307 GB United Kingdom Mobile - H3G 0.0074
447409 GB United Kingdom Mobile - Orange 0.0083
447107 GB United Kingdom Mobile - Telefonica 0.0072
but mostly to audibly prepare the call handler as they deal with other customers. I think I may have found (a/the) solution however - set up a Queue.