[Asterisk-Users] IConnectHere and DTMF

[Ben Clark]

Does anyone know if asterisk can be configured to accept DTMF inbound on SIP calls?
> I am just using the IConnectHere service directly with
> asterisk (no ATA186).  When calling asterisk by
> dialing my IConnectHere DID number from my cell phone
> I can here the tones corectly, but asterisk doesn't
> seem to regonize them for the IVR menus.  Maybe
> asterisk is not able to understand inband DTMF on
> incoming / outgoing SIP calls?
--- John Todd <jtodd at loligo.com> wrote:
> In short: I know of no way to get DTMF working
> through a Cisco 
> ATA-186 (SIP) and Asterisk to anything other than
> the X100P port.
> 
> What kind of equipment are you using?  I've tried my
> Cisco ATA-186 
> (v2.15) in various settings (in-band, out-of-band,
> out-of-band-with 
> negotiation) and none of them work.  Asterisk can
> see the DTMF, but 
> can't send it. (actually, after the last CVS I did
> earlier yesterday, 
> I no longer even see the "DTMF pending" and "DTMF
> sent" messages on 
> the console, but I hear the static when I hit keys
> on the ATA-186 
> phone.  I now see "Difference is 1928, ms is 40792"
> type messages 
> when I hit DTMF on the ATA)
> 
> I've experimented with my own Cisco gateway
> (12.2(12) on a 3640) and 
> have been unable to get DTMF to work, despite
> various settings on the 
> ATA-186 and configuration options (dtmf-relay) on
> the 3640.
> 
> Whenever I send DTMF from PSTN -> 3640 -> * -> ATA,
> I can see this 
> message on the console:
> 
> NOTICE[27662]: File rtp.c, Line 300 (ast_rtp_read):
> Unknown RTP codec 
> 19 received
> 
> I can hear the DTMF, but it doesn't do anything
> (doesn't work in any 
> menus I create.)
> 
> Yes, I've followed the instructions on 
> http://www.djernes.org/~shawn/ata186.htm  - no luck.
>  I've even 
> followed the instructions on 
>
http://corp.deltathree.com/productsandservices/devices/ATAConfiguration.doc
> 
> with no success.
> 
> JT
> 
> 
> >I have IConnectHere inbound and outbound service
> working with great 
> >sound quality; however, I am unable to send or
> receive DTMF tones 
> >when calling inbound or outbound.  Does anyone know
> what 
> >IConnectHere supports in regards to DTMF and how
> asterisk can be 
> >configured to cooperate?
> >
> >Also, I constantly receive the following messages
> on the console 
> >since using the SIP register feature.  The timeout
> numbers start at 
> >1 and go up.
> >
> >NOTICE[13326]: File chan_sip.c, Line 1920
> (transmit_register): 
> >Scheduled a timeout # 121
> >NOTICE[13326]: File chan_sip.c, Line 2894
> (handle_response): 
> >Registration successful
> >NOTICE[13326]: File chan_sip.c, Line 2895
> (handle_response): 
> >Cancelling timeout 121
> >NOTICE[13326]: File chan_sip.c, Line 1920
> (transmit_register): 
> >Scheduled a timeout # 123
> >NOTICE[13326]: File chan_sip.c, Line 2894
> (handle_response): 
> >Registration successful
> >NOTICE[13326]: File chan_sip.c, Line 2895
> (hndle_response): 
> >Cancelling timeout 123



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