I have a setup where there are occasional problems with attended transfers. I have already checked the devices as well as the relevant DTMF modes (SIP INFO and rfc2833). I could not find any problems here.

The setup is a follows:

The front desk (F) accepts calls from customers (C). In some cases F needs to transfer C to a specific department (D). If D cannot handle the problem, D tries to transfer to a specialist S. The problem sometimes occurs when D tries an attended transfer for C to S. The description I
have so far is that Asterisk sometimes does not seem to accept DTMF signals.

Is it conceivable that that this is a re-INVITE/directmedia problem?

Since I have only limited access to the system, I’d like to rule out some causes. I am also not sure if the transfer from F to D is handled via Asterisk using DTMF signalling, or whether F
uses the SIP phone’s capabilities for the transfer, so that Asterisk might already not know anything about D. Except for C, everybody is inside the same subnet.

jg

Hello list,

‘Just wondering if anyone can point to a very light-weight and easy to incorporate into Asterisk (v. 11.x) to handle a minimal set of responses, like:
0 – 9
yes
no
(maybe * and # for some people)

The idea is that within an IVR menu, the caller could respond by speaking to the typical IVR options, like:

For Archie, press or say 1 now
For Veronica, press or say 2 now
For Jughead, press or say 3 now
(etc.)

You have selected option 2 for Veronica, press 1 or say “yes” if this is correct.

If a voice response was received (not a DTMF key press) indeterminate, some status would be useful (beyond just a timeout).

It would be great if this was simple to code into the dialplan, much like like the current background/wait model for keypresses. Low cost or free would be nice too!

Thanks for any suggestions.

Hello,

I notice that it takes 4 to 6 seconds between someone pressing a cipher and Asterisk continuing inside the dialplan. How come ???

Taken from verbose logfile :

(attempt 1)
[Jun 11 15:29:25] DTMF[18549] channel.c: DTMF begin ’1′ received on SIP/SipAgenT01-00001eb0
[Jun 11 15:29:25] DTMF[18549] channel.c: DTMF begin ignored ’1′ on SIP/SipAgenT01-00001eb0
[Jun 11 15:29:25] DTMF[18549] channel.c: DTMF end ’1′ received on SIP/SipAgenT01-00001eb0, duration 180 ms
[Jun 11 15:29:25] DTMF[18549] channel.c: DTMF end passthrough ’1′ on SIP/SipAgenT01-00001eb0

[Jun 11 15:29:30] VERBOSE[18549] pbx.c: [Jun 11 15:29:30] == CDR
updated on SIP/SipAgenT01-00001eb0
[Jun 11 15:29:30] VERBOSE[18549] pbx.c: [Jun 11 15:29:30]

Hi,

I am receiving DTMF without any reason after call establishment.

The log as follows, and I suspect something related to directmedia,
[May 17 00:33:35] verbose[4238] app_dial.c: — SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48
[May 17 00:33:35] VERBOSE[4238] app_dial.c: — SIP/MyTrunk-000a4b49
answered SIP/MAN-000a4b48
[May 17 00:33:35] DTMF[4238] channel.c: DTMF end ‘*’ received on SIP/MyTrunk-000a4b49, duration 0 ms
[May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin ‘*’
on SIP/MyTrunk-000a4b49
[May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough ‘*’ on SIP/MyTrunk-000a4b49
[May 17 00:33:36] DTMF[4238] channel.c: DTMF end ’8′ received on SIP/MyTrunk-000a4b49, duration 0 ms
[May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin ’8′
on SIP/MyTrunk-000a4b49
[May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough ’8′ on SIP/MyTrunk-000a4b49
[May 17 00:33:36] DTMF[4104] channel.c: DTMF end ’8′ received on SIP/MAN-000a4af0, duration 100 ms
[May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of ’8′ with duration 100 queued on SIP/MAN-000a4af0
[May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of ’8′ queued on SIP/MAN-000a4af0
[May 17 00:33:37] DTMF[4234] channel.c: DTMF end ’1′ received on SIP/MAN-000a4b41, duration 100 ms
[May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of ’1′ with duration 100 queued on SIP/MAN-000a4b41
[May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of ’1′ queued on SIP/MAN-000a4b41
[May 17 00:33:55] VERBOSE[4106] pbx.c: == Spawn extension
(sip-trunk-inbound, 2127773456, 1) exited non-zero on ‘SIP/MyTrunk-000a4af3′
[May 17 00:33:56] VERBOSE[4136] pbx.c: — Executing [h@trunk-outbound:1]
NoOp(“SIP/MAN-000a4b09″, “16″) in new stack
[May 17 00:33:56] VERBOSE[4136] pbx.c: == Spawn extension
(trunk-outbound, 777787457712, 2) exited non-zero on ‘SIP/MAN-000a4b09′

Is this some thing related to SIP RE-INVITE?

Thanks.

Hi, I’ve noticed on asterisk 1.8.18 I’m hearing the blip of ‘#’ DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this?
The DTMF is coming in through rfc2833 and not inband.

Thanks.

– James