[asterisk-users] DTMF transmission problem

[Noah Engelberth]

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Noah Engelberth
> Sent: Thursday, August 02, 2012 12:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] DTMF transmission problem
> 
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> > bounces at lists.digium.com] On Behalf Of Shaun Ruffell
> > Sent: Thursday, August 02, 2012 11:06 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] DTMF transmission problem
> >
> > On Thu, Aug 02, 2012 at 12:45:28PM +0000, Noah Engelberth wrote:
> > > I am having difficulties with customer-bound DTMF being very short &
> > > clipped off (and basically unusable, as systems on the customer side
> > > aren't recognizing the DTMF digits, and I can barely tell that DTMF
> > > is there when I listen on a handset).
> > >
> > > My system set up as follows:
> > >
> > > PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE
> >
> > [snip]
> >
> > > ... Vocal call  quality is fine, DTMF is fine from the customer to
> > > the PSTN, but DTMF from the PSTN to the customer isn't ...
> >
> >  [snip]
> >
> > > The same symptoms persist whether the PSTN or the CPE initiate the call.
> >
> > What is the dtmf mode of Metaswitch in the above diagram? Is it
> > possible that it's muting the DTMF and then not generating the
> > corresponding DTMF event messages?  Everytime I've seen "clipped"
> > DTMF in the past it was due to imperfect muting at the PSTN -> SIP
> interface.
> 
> According to the gentleman that manages the Metaswitch, it's set to allow
> for either in or out of band dtmf.  Based on the packet trace, the packets are
> coming across as RFC 2833 RTP events.  Aside from the very first digit, which
> Wireshark shows as 7 "RTP Event" packets and 3 "RTP Event (end)" packets,
> all the other ones on my test call came across as 8 "RTP Event" packets and 3
> "RTP Event (end)" packets.  All of the RTP Event packets are in sequence for
> the call's RTP stream.
> 
> Also, when I'm monitoring in Asterisk, if I configure logger.conf to output
> DTMF events into the console, Asterisk is recognizing the DTMF:
> 
> [Aug  2 12:25:25] DTMF[19319]: channel.c:4136 __ast_read: DTMF begin '4'
> received on SIP/PSTN-SIP-PEER [Aug  2 12:25:25] DTMF[19319]:
> channel.c:4146 __ast_read: DTMF begin passthrough '4' on SIP/ PSTN-SIP-
> PEER [Aug  2 12:25:25] DTMF[19319]: channel.c:4051 __ast_read: DTMF end
> '4' received on SIP/ PSTN-SIP-PEER, duration 280 ms [Aug  2 12:25:25]
> DTMF[19319]: channel.c:4091 __ast_read: DTMF end accepted with begin '4'
> on SIP/ PSTN-SIP-PEER [Aug  2 12:25:25] DTMF[19319]: channel.c:4120
> __ast_read: DTMF end passthrough '4' on SIP/ PSTN-SIP-PEER
> 

Additional information I discovered after my previous reply:

I have a separate Asterisk VM instance (in all other ways the same implementation as above) that is running an IVR.  This instance has no issues with inbound DTMF within the IVR, but does exhibit the same symptoms for DTMF when bridged through to an IAX2 peer with the same settings as the first Asterisk VM.  On the second Asterisk (with the IVR), DTMF to my Cisco/Linksys SPA942 SIP phones works properly, but not to the IAX or SIP ATAs that I am using (the same ones I'm having problems with on the first Asterisk).  All of the live customers on the first Asterisk are ATAs, so I don't know as of this time whether or not SPA phones are working correctly on the first server, though it's reasonable to assume they are.

In addition, calls from an SPA942 phone to the IAX or SIP ATAs are also not transmitting DTMF to the ATA device's endpoint.  DTMF from the ATA device's endpoint to the SPA942 is working correctly, as is both directions of voice audio.

> >
> > You should be able to take a packet trace on the interface of the
> > Asterisk server communicating with the Metaswitch to determine whether
> > the problem first appears at the switch or in your Asterisk server.
> >
> > Cheers,
> > Shaun
> >
> > --
> > Shaun Ruffell
> > Digium, Inc. | Linux Kernel Developer
> > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
> > www.digium.com & www.asterisk.org
> >
> > --