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