vicidial.org

Hi,

We are unable to send the DTMF digits/menu options (like press1 for this, press2 for this, etc) from softphone while login as an agent. But we are able to send the DTMF digits without login as an agent in dialer, we can pass the digit directly from softphone (using carrier prefix).

Below are the configuration details of our server.

Installation Method: Manually from Ray Solomon site.
RAM: 32 GB
CPU: Total no of CPU: 24
cat /usr/src/astguiclient/trunk/version
2.14b0.5
Asterisk 13.29.2

Below are the Asterisk CLI logs.

Pressed-1
[Mar 6 18:13:44] DTMF[5580][C-00000ec4]: channel.c:4152 __ast_read: DTMF begin '1' received on SIP/3301-000003a6
[Mar 6 18:13:44] DTMF[5580][C-00000ec4]: channel.c:4163 __ast_read: DTMF begin passthrough '1' on SIP/3301-000003a6
[Mar 6 18:13:44] DTMF[5580][C-00000ec4]: channel.c:4066 __ast_read: DTMF end '1' received on SIP/3301-000003a6, duration 100 ms
[Mar 6 18:13:44] DTMF[5580][C-00000ec4]: channel.c:4107 __ast_read: DTMF end accepted with begin '1' on SIP/3301-000003a6
[Mar 6 18:13:44] DTMF[5580][C-00000ec4]: channel.c:4136 __ast_read: DTMF end passthrough '1' on SIP/3301-000003a6
EurusLV01*CLI>
EurusLV01*CLI>

Pressed-2
[Mar 6 18:13:49] DTMF[5580][C-00000ec4]: channel.c:4152 __ast_read: DTMF begin '2' received on SIP/3301-000003a6
[Mar 6 18:13:49] DTMF[5580][C-00000ec4]: channel.c:4163 __ast_read: DTMF begin passthrough '2' on SIP/3301-000003a6
[Mar 6 18:13:49] DTMF[5580][C-00000ec4]: channel.c:4066 __ast_read: DTMF end '2' received on SIP/3301-000003a6, duration 80 ms
[Mar 6 18:13:49] DTMF[5580][C-00000ec4]: channel.c:4107 __ast_read: DTMF end accepted with begin '2' on SIP/3301-000003a6
[Mar 6 18:13:49] DTMF[5580][C-00000ec4]: channel.c:4122 __ast_read: DTMF end '2' detected to have actual duration 53 on the wire, emulation will be triggered on SIP/3301-000003a6
[Mar 6 18:13:49] DTMF[5580][C-00000ec4]: channel.c:4129 __ast_read: DTMF end '2' has duration 53 but want minimum 80, emulating on SIP/3301-000003a6
[Mar 6 18:13:49] DTMF[5580][C-00000ec4]: channel.c:4186 __ast_read: DTMF end emulation of '2' queued on SIP/3301-000003a6

Please advise...Thanks...!!!

It looks like you are sending DTMF but it's not working on the SIP remote side. DTMF issues are often caused by codec issues, if you can try a different codec or else you will need to take this issue up with your SIP provider.

Thanks Vince for the reply and sorry for being late, I was stuck somewhere.

I'll check and let you know the status.

1) Why are these agents not using the DTMF field on the agent screen? (Not being a smartass here, knowing the reason behind this may be useful in someone providing an answer)

2) Agent phones use a DTMFMode setting to allow asterisk to capture the DTMF from the agent. Then the meetme room loses it completely because meetme doesn't transfer dtmf, only sound. However: if the carrier can catch the DTMF through sound, perhaps based on the DTMFmode of the Carrier's SIP account configuration ... play with the dtmf settings.

https://www.voip-info.org/asterisk-sip-dtmfmode/

The setting is in each context of the sip.conf and/or sip-vicidial.conf (Admin->Carriers->SIP Account for each and for agent phones uses sip.conf setting by default unless overridden)