Custom ringtone for transferred call

FreePBX Configuration
1

Good Morning Colleagues
I hope you are doing well . I have installed FreePBX V16 and having 3 extensions where each extension is configured on a Microsip softphone.

All 3 extensions related to one queue .

In some situations , When a call received by Agent 1 (extension 1) , It may be required to be transferred to Agent 2 (extension 2) .

When the call is being transferred , The caller hears normal/classic ring tone .

May i know how to change this ring tone and select any custom ring tone that i need instead of the normal/classic ring tone ?

Best Regards

2

You can try Extension → Advanced → Asterisk Dial Options and append in parentheses the ringing tone you want the caller to hear.

So for example, if your settings currently looks like this HhTtr change it to HhTtr(tone)

If you want Music on Hold instead, then change it to HhTtm(class)

Source:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Application_Dial

3

Good Afternoon PitzKey
Thx for your replay .Indeed , I need to add MOH instead of the classic ring . The MOH i need to add is “02” as per the attached.

I configure “Asterisk Dial Options” as per the attached but i still hear the classic ring.

Kindly let me know how to fix the issue to hear MOH “02” instead of classic ring.

Thx in advance.

Best Regards

Asterisk Dial Options
Asterisk Dial Options771×132 16 KB

Music on Hold
Music on Hold812×271 28.2 KB

4

Please provide a call trace via pastebin Providing Great Debug - Support Services - Documentation

Use pastebin.com as pastebin.freepbx.org is down

5

How is the transfer requested?

  • blind transfer feature code
  • attended transfer feature code
  • register recall on analogue line
  • SIP attended transfer (including cases where the phone uses this for blind transfers)
  • SIP blind transfer
  • other.

Please confirm that this is an answered call that is being transferred.

6

Good Afternoon David
The call waiting in Queue has been sent to Agent 1 then agent 1 click on Transfer button on Microsip softphone , write the extension number of Agent 2 (102) then click transfer to transfer the call to agent 2 as in below snapshot.

What is required that while call is being transferred from Agent 1 to Agent 2 , I need the caller to hear MOH instead of the classic ring.

Call Transfer - Microsip

Best Regards

7

Good Afternoon PitzKey
I have tested again but it didn’t work in the required scenario. It worked only when agent 1 needs to call any extension (either agent 2 or agent 3) . At this point , Agent 1 can hear the MOH.

What is required is the below:

The call waiting in Queue has been sent to Agent 1 then agent 1 click on Transfer button on Microsip softphone , write the extension number of Agent 2 (102) then click transfer to transfer the call to agent 2 as in below snapshot.

What is required that while call is being transferred from Agent 1 to Agent 2 , I need the caller to hear MOH instead of the classic ring.

Call Transfer - Microsip

Best Regards

8

Where are your logs?

9

What ring back tone does the person initiating the transfer between starting and completing the transfer?

What ring back tone would the caller have heard when originally called in?

Again pleas confirm that the caller has been answered.

What version of Asterisk are you using?

Does the original caller hear music on hold between the start of the transfer sequence and its completion?

I’m currently assuming this is a native SIP attended transfer.

(I’m guessing that either the hold to set up the new music on hold is conflicting with the unhold when the caller is reconnected, or the callee channel doesn’t remember it put someone on hold, so dosen’t repeat the hold request when connected to the original caller. Both of these are likely to be in Asterisk and, therefore, difficult to resolve.)

10

Hello David
Kindly find my below replay:

  • The person initiating the transfer hears nothing between starting and completing the transfer (It is blind transfer)
  • The caller heard the Announcement message configured for the IVR when he originally called in.
  • The caller has been answered.
  • Asterisk version is 18.16.0.
  • The original caller didn’t put on hold . He can talk to the original agent while this agent is transferring him . Once Blind transfer done , The other agent phone’s ring and the original callers hears the classic ring back tone.

Best Regards

11

This sounds more like an attended transfer being completed without listening, than a blind transfer, but I can’t find any detailed documentation on MicroSIP transfers.

12

Well Noted and thanks for trying to help David . By the way , DO you recommend another free softphone better than Microsip that you have personally used ?

Best Regards

13

That is unusual. Normally, even for a blind transfer, the initial caller would be put on hold when the transfer button is pressed.

We definitely need the logs, and may need the protocol logging, to understand what the phone is actually doing.

14

Hello David
I tried to generate the logs using the below document but it gives error maybe because " pastebin.freepbx.org" is down as PitzKey said so may i know how to use " pastebin.com" instead ?

https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs-PartII

15

Hello David
I generate the logs and paste it manually on pastebin . Here is the below link:

Best Regards

16

It does seem to be a genuine blind transfer:

1151. [2023-03-14 17:17:35] VERBOSE[119910][C-00000017] app_macro.c: Channel 'Local/[email protected];2' jumping out of macro 'dial-one'
1152. [2023-03-14 17:17:35] VERBOSE[119910][C-00000017] app_macro.c: Channel 'Local/[email protected];2' jumping out of macro 'exten-vm'
1153. [2023-03-14 17:17:35] VERBOSE[119910][C-00000017] pbx.c: Executing [[email protected]:1] GotoIf("Local/[email protected];2", "0?ext-local,*102,1") in new stack

and it isn’t requesting music on hold when dialling PJSIP/102

1339. [2023-03-14 17:17:35] VERBOSE[119910][C-00000017] pbx.c: Executing [[email protected]:56] Dial("Local/[email protected];2", "PJSIP/102/sip:[email protected]:1026;ob,,HhtrM(auto-blkvm)Ib(func-apply-sipheaders^s^1)") in new stack

The main obvious difference is that it is using context from-internal-xfer, not from-internal, or from-queue, which suggests this might be a FreePBX issue rather than an Asterisk one (I’d only expect Asterisk to get into trouble on attended transfers).

There is no m(0… anywhere in the trace, so I assume that from-queue doesn’t honour this either.

Especially without an example of this working, I think it is going to take me too long to trawl the code to see why it is not getting set on this path. It is always possible that this is intended behaviour, for some reason. It is possible that there was a policy decision that dial options should not change during a call, but should reflect the initial DID/dialled number used.

It does, however, look to me as though the only place this would get set in the current code might be here:

128. [2023-03-14 17:16:10] VERBOSE[119714][C-00000017] pbx.c: Executing [[email protected]:5] ExecIf("PJSIP/10000-0000003f", "1?Set(_DIAL_OPTIONS=HhTtrM(auto-blkvm))") in new stack

which suggests that setting it for the queue might affect the transfer, always assuming that setting it for the queue is compatible with what you want to achieve.

17

Thanks David so much for your investigation and trying to help . I will try to find workaround.

By the way , Is Microsip good softphone or do you recommend better free softphone ?

Best Regards

18

I have no experience of the current generation of soft phones. The older ones I used all had limitations and bugs, and I only used them to make up numbers when testing call scenarios, never for production use.

19

Thx David for your feedback

20

Looks like the dial options on the extension level is for calls placed from that extension.

You can try setting it globally under advanced settings