Feature codes with Digium phones

Asterisk Asterisk Support
1

I’m trying to enable on-demand recording using a *3 feature code. I’m trying to follow this guide: http://www.fosslc.org/drupal/node/643

However, when on a call, and I hit “*3”, asterisk just sends those DTMF tones:

[quote][Dec 2 11:17:26] DTMF[8123]: channel.c:4194 __ast_read: DTMF begin ‘’ received on SIP/22124-00000004
[Dec 2 11:17:26] DTMF[8123]: channel.c:4204 __ast_read: DTMF begin passthrough '’ on SIP/22124-00000004
[Dec 2 11:17:26] DTMF[8123]: channel.c:4109 __ast_read: DTMF end ‘’ received on SIP/22124-00000004, duration 170 ms
[Dec 2 11:17:26] DTMF[8123]: channel.c:4149 __ast_read: DTMF end accepted with begin '’ on SIP/22124-00000004
[Dec 2 11:17:26] DTMF[8123]: channel.c:4178 __ast_read: DTMF end passthrough ‘*’ on SIP/22124-00000004
[Dec 2 11:17:27] DTMF[8123]: channel.c:4194 __ast_read: DTMF begin ‘3’ received on SIP/22124-00000004
[Dec 2 11:17:27] DTMF[8123]: channel.c:4204 __ast_read: DTMF begin passthrough ‘3’ on SIP/22124-00000004
[Dec 2 11:17:27] DTMF[8123]: channel.c:4109 __ast_read: DTMF end ‘3’ received on SIP/22124-00000004, duration 120 ms
[Dec 2 11:17:27] DTMF[8123]: channel.c:4149 __ast_read: DTMF end accepted with begin ‘3’ on SIP/22124-00000004
[Dec 2 11:17:27] DTMF[8123]: channel.c:4178 __ast_read: DTMF end passthrough ‘3’ on SIP/22124-00000004[/quote]

I have the feature turned on:

[quote]PBXTEST*CLI> features show
Builtin Feature Default Current

Pickup *8 *8
Blind Transfer # CC
Attended Transfer
One Touch Monitor *3
Disconnect Call * B
Park Call
One Touch MixMonitor [/quote]

And I have [quote][globals]
DYNAMIC_FEATURES=>automon[/quote] in the extensions.conf.

The last detail I believe is the digit mapping on the phones. Right now it’s:

Does anyone know what I’m missing to make this feature work?

2

Howdy,

That DTMF debug there shows the *3 being received, so the phone’s sending it. The phone’s dial / digit plan only affects off-hook dialing. Once you’re on a call, it’s not in play.

Check your Dial application in your dialplan to see if the person dialing and sending the DTMF has the X flag - if what you want is automixmon.

Cheers

3

I altered the features.conf to be as follows:

[featuremap] ;blindxfer => #1 ; Blind transfer ;disconnect => *0 ; Disconnect automixmon => *3 ; One Touch Record ;atxfer => *2 ; Attended transfer

And proof it’s setup:

[code]PBXTEST*CLI> features show
Builtin Feature Default Current

Pickup *8 *8
Blind Transfer # CC
Attended Transfer
One Touch Monitor
Disconnect Call * B
Park Call
One Touch MixMonitor *3 [/code]

For the extensions.conf I have the following Dial() steps with the “X”:

TRUNKASP=SIP/sip-airespring-long [...] exten => _41NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _41NXXNXXXXXX,n,Dial(${TRUNKASP}/${EXTEN:1},,X) exten => _41NXXNXXXXXX,n,Hangup exten => _40111XXXXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _40111XXXXXXXXX.,n,Dial(${TRUNKASP}/${EXTEN:4},,X) exten => _40111XXXXXXXXX.,n,Hangup() exten => _4011X.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _4011X.,n,Dial(${TRUNKASP}/${EXTEN:1},,X) exten => _4011X.,n,Hangup()

However, the same behavor happens. I have an established call, but dialing “*3” just results in those DTMF tones being played on the call.

[Dec 3 12:35:34] == Using SIP RTP CoS mark 5 [Dec 3 12:35:34] -- Executing [1111@default:1] Dial("SIP/22124-00000007", "SIP/sip-airespring-long/12169040991") in new stack [Dec 3 12:35:34] == Using SIP RTP CoS mark 5 [Dec 3 12:35:34] -- Called SIP/sip-airespring-long/12169040991 [Dec 3 12:35:35] -- SIP/sip-airespring-long-00000008 is ringing [Dec 3 12:35:35] -- SIP/sip-airespring-long-00000008 is making progress passing it to SIP/22124-00000007 [Dec 3 12:35:41] -- SIP/sip-airespring-long-00000008 answered SIP/22124-00000007 [Dec 3 12:35:43] DTMF[21752]: channel.c:4194 __ast_read: DTMF begin '*' received on SIP/22124-00000007 [Dec 3 12:35:43] DTMF[21752]: channel.c:4204 __ast_read: DTMF begin passthrough '*' on SIP/22124-00000007 [Dec 3 12:35:44] DTMF[21752]: channel.c:4109 __ast_read: DTMF end '*' received on SIP/22124-00000007, duration 220 ms [Dec 3 12:35:44] DTMF[21752]: channel.c:4149 __ast_read: DTMF end accepted with begin '*' on SIP/22124-00000007 [Dec 3 12:35:44] DTMF[21752]: channel.c:4178 __ast_read: DTMF end passthrough '*' on SIP/22124-00000007 [Dec 3 12:35:44] DTMF[21752]: channel.c:4194 __ast_read: DTMF begin '3' received on SIP/22124-00000007 [Dec 3 12:35:44] DTMF[21752]: channel.c:4204 __ast_read: DTMF begin passthrough '3' on SIP/22124-00000007 [Dec 3 12:35:44] DTMF[21752]: channel.c:4109 __ast_read: DTMF end '3' received on SIP/22124-00000007, duration 190 ms [Dec 3 12:35:44] DTMF[21752]: channel.c:4149 __ast_read: DTMF end accepted with begin '3' on SIP/22124-00000007 [Dec 3 12:35:44] DTMF[21752]: channel.c:4178 __ast_read: DTMF end passthrough '3' on SIP/22124-00000007 [Dec 3 12:35:49] -- Executing [h@default:1] AGI("SIP/22124-00000007", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----15-----8") in new stack [Dec 3 12:35:49] -- <SIP/22124-00000007>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----15-----8 completed, returning 0

Any thoughts on what I’m missing to make this work?

4

This is still an issue for me. Does anyone have any ideas or solution in mind?

Thank you in advance.

5

I found the solution to this. To test, I was using a speeddial button that had a pre-configured Dial() step, that wasn’t using the flags. Testing using the correct dial step did in fact catch the *3 code. It works.