I believe the TIMEOUT() function and apps only work once you are in an IVR or other dialplan application which waits for digits. On DAHDI channels I think you have to modify the source code if you want to change the timeout when dialing from a dialtone.
-----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Justin Killen Sent: Wednesday, July 10, 2013 12:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay Okay, after enabling DTMF logging, what I see is a handset being picked up, 7 digits being pressed in 4 seconds, and then 3 seconds input is determined to be done and the call is processed (to the catch-all 'bad-number'). What I don't understand is that if the digit timeout is set to 5, then why do the calls attempt to process only after 3 seconds? Following is output from the call log (I have the DEBUG output too if that is needed). [2013-07-10 09:22:37] VERBOSE[12753][C-0002ec16] sig_analog.c: -- Starting simple switch on 'DAHDI/96-1' [2013-07-10 09:22:38] DTMF[12753][C-0002ec16] channel.c: DTMF begin '1' received on DAHDI/96-1 [2013-07-10 09:22:38] DTMF[12753][C-0002ec16] channel.c: DTMF begin ignored '1' on DAHDI/96-1 [2013-07-10 09:22:38] DTMF[12753][C-0002ec16] channel.c: DTMF end '1' received on DAHDI/96-1, duration 89 ms [2013-07-10 09:22:38] DTMF[12753][C-0002ec16] channel.c: DTMF end passthrough '1' on DAHDI/96-1 [2013-07-10 09:22:38] DTMF[12753][C-0002ec16] channel.c: DTMF begin '9' received on DAHDI/96-1 [2013-07-10 09:22:38] DTMF[12753][C-0002ec16] channel.c: DTMF begin ignored '9' on DAHDI/96-1 [2013-07-10 09:22:38] DTMF[12753][C-0002ec16] channel.c: DTMF end '9' received on DAHDI/96-1, duration 89 ms [2013-07-10 09:22:38] DTMF[12753][C-0002ec16] channel.c: DTMF end passthrough '9' on DAHDI/96-1 [2013-07-10 09:22:38] DTMF[12753][C-0002ec16] channel.c: DTMF begin '0' received on DAHDI/96-1 [2013-07-10 09:22:38] DTMF[12753][C-0002ec16] channel.c: DTMF begin ignored '0' on DAHDI/96-1 [2013-07-10 09:22:38] DTMF[12753][C-0002ec16] channel.c: DTMF end '0' received on DAHDI/96-1, duration 89 ms [2013-07-10 09:22:38] DTMF[12753][C-0002ec16] channel.c: DTMF end passthrough '0' on DAHDI/96-1 [2013-07-10 09:22:39] DTMF[12753][C-0002ec16] channel.c: DTMF begin '9' received on DAHDI/96-1 [2013-07-10 09:22:39] DTMF[12753][C-0002ec16] channel.c: DTMF begin ignored '9' on DAHDI/96-1 [2013-07-10 09:22:39] DTMF[12753][C-0002ec16] channel.c: DTMF end '9' received on DAHDI/96-1, duration 140 ms [2013-07-10 09:22:39] DTMF[12753][C-0002ec16] channel.c: DTMF end passthrough '9' on DAHDI/96-1 [2013-07-10 09:22:40] DTMF[12753][C-0002ec16] channel.c: DTMF begin '9' received on DAHDI/96-1 [2013-07-10 09:22:40] DTMF[12753][C-0002ec16] channel.c: DTMF begin ignored '9' on DAHDI/96-1 [2013-07-10 09:22:40] DTMF[12753][C-0002ec16] channel.c: DTMF end '9' received on DAHDI/96-1, duration 102 ms [2013-07-10 09:22:40] DTMF[12753][C-0002ec16] channel.c: DTMF end passthrough '9' on DAHDI/96-1 [2013-07-10 09:22:40] DTMF[12753][C-0002ec16] channel.c: DTMF begin '9' received on DAHDI/96-1 [2013-07-10 09:22:40] DTMF[12753][C-0002ec16] channel.c: DTMF begin ignored '9' on DAHDI/96-1 [2013-07-10 09:22:40] DTMF[12753][C-0002ec16] channel.c: DTMF end '9' received on DAHDI/96-1, duration 102 ms [2013-07-10 09:22:40] DTMF[12753][C-0002ec16] channel.c: DTMF end passthrough '9' on DAHDI/96-1 [2013-07-10 09:22:41] DTMF[12753][C-0002ec16] channel.c: DTMF begin '6' received on DAHDI/96-1 [2013-07-10 09:22:41] DTMF[12753][C-0002ec16] channel.c: DTMF begin ignored '6' on DAHDI/96-1 [2013-07-10 09:22:41] DTMF[12753][C-0002ec16] channel.c: DTMF end '6' received on DAHDI/96-1, duration 127 ms [2013-07-10 09:22:41] DTMF[12753][C-0002ec16] channel.c: DTMF end passthrough '6' on DAHDI/96-1 [2013-07-10 09:22:44] VERBOSE[12753][C-0002ec16] pbx.c: -- Executing [1909996@from-internal:1] NoOp("DAHDI/96-1", "bad-number, timeouts: absolute: 0 digit: 5.000 response: 10.000") in new stack [2013-07-10 09:22:44] VERBOSE[12753][C-0002ec16] pbx.c: -- Executing [1909996@from-internal:2] ResetCDR("DAHDI/96-1", "") in new stack [2013-07-10 09:22:44] VERBOSE[12753][C-0002ec16] pbx.c: -- Executing [1909996@from-internal:3] NoCDR("DAHDI/96-1", "") in new stack [2013-07-10 09:22:44] VERBOSE[12753][C-0002ec16] pbx.c: -- Executing [1909996@from-internal:4] Progress("DAHDI/96-1", "") in new stack [2013-07-10 09:22:44] VERBOSE[12753][C-0002ec16] pbx.c: -- Executing [1909996@from-internal:5] Wait("DAHDI/96-1", "1") in new stack [2013-07-10 09:22:45] VERBOSE[12753][C-0002ec16] pbx.c: -- Executing [1909996@from-internal:6] Progress("DAHDI/96-1", "") in new stack [2013-07-10 09:22:45] VERBOSE[12753][C-0002ec16] pbx.c: -- Executing [1909996@from-internal:7] Playback("DAHDI/96-1", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack [2013-07-10 09:22:45] VERBOSE[12753][C-0002ec16] file.c: -- <DAHDI/96-1> Playing 'silence/1.ulaw' (language 'en') [2013-07-10 09:22:46] VERBOSE[12753][C-0002ec16] pbx.c: == Spawn extension (from-internal, 1909996, 7) exited non-zero on 'DAHDI/96-1' [2013-07-10 09:22:46] VERBOSE[12753][C-0002ec16] pbx.c: -- Executing [h@from-internal:1] Hangup("DAHDI/96-1", "") in new stack [2013-07-10 09:22:46] VERBOSE[12753][C-0002ec16] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'DAHDI/96-1' [2013-07-10 09:22:46] VERBOSE[12753][C-0002ec16] sig_analog.c: -- Hanging up on 'DAHDI/96-1' [2013-07-10 09:22:46] VERBOSE[12753][C-0002ec16] chan_dahdi.c: -- Hungup 'DAHDI/96-1' -Justin -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of jg Sent: Monday, July 08, 2013 11:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay > The channel banks are Adtran TA-624's using ESF/B8ZS. When a handset is > picked up, I can see the offhook in the asterisk console, so it looks that > the channel is immediately connected through the channel bank (not delayed > until after digits are dialed), so it looks that overlap dialing isn't a > factor and that asterisk has complete control. This also means that you should see the digits as they are dialed. When something times out you should also see a message why there was a timeout. I am using ISDN for PSTN connections and where I live there must be some kind of overlap dialing enabled, otherwise P2P configurations don't work. With current DAHDI drivers I no longer need special settings to make things work (Wanpipe/Woomera was different), so I guess overlap dialing is enabled. Some SIP phones distinguish between "Overlap Dialing" and "Automatic Dialing", so your channel bank might also have something like an Automatic Dialing option with some timing value. > As for options in chan_dahdi.conf, I simply can't find any that relate to > this problem. I have looked at the page here: > http://www.voip-info.org/wiki/view/chan_dahdi.conf and the closest thing I > can find is 'ringtimeout' which is obviously not what I want. I would expect > to see something like 'dialtimeout' or 'interdigittimeout'. There is an "overlap" option in configs/chan_dahdi.conf.sample. I am currently assembling an Asterisk box that has 48+2 analog channels (+ SIP + ISDN). If your problem doesn't go away I could tell next week what my system is doing. jg -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
