[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 1:10 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 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
> > >
> > > --
> 
> --

At the risk of answering myself --

Found that on calls Asterisk was bridging together and not "hearing" the DTMF, it was working normally.  On calls that Asterisk was still "hearing" the DTMF, it was being clipped.  It seems that Asterisk was involved in relaying the DTMF on calls to IAX endpoints as well as calls to SIP endpoints on a different network from the Asterisk customer-facing network interface (in all cases, Asterisk was locally bridging calls, as I have directmedia=no set for all SIP peers and the MetaSwitch is on a different interface/network than the customers anyway).  When I inserted the line "dtmfmode = inband" for the problematic SIP peers, DTMF began working normally.  A little frustrating, in that the peer should have been requesting InBand DTMF based on its config, and even changing the peer's config to RFC 2833 DTMF wasn't helping things, but at least things appear to be working now.

Thank you,

Noah Engelberth
MetaLINK Technologies