[asterisk-users] Subscribe to Local channel status
Hi, Is it possible to assign hint extension to Local channel? Something like this: exten = 555,hint,Local/123123123@my-context The purpose is to subscribe to this channel state from SIP-phone. I know that queues can track Local channel status, however I could not find any information regarding using Local channel in hints. -- Best regards, Grigoriy -- _ -- 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
Re: [asterisk-users] asterisk-users Digest, Vol 108, Issue 14
Unsubscribe Elvin G. Nodalo -Original Message- From: asterisk-users-requ...@lists.digium.com Sent: 7/10/2013 1:00 AM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 108, Issue 14 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. analog phone digit delay (Justin Killen) 2. Re: analog phone digit delay (jg) 3. Re: analog phone digit delay (Justin Killen) 4. Re: analog phone digit delay (jg) 5. Re: analog phone digit delay (Steve Edwards) 6. Re: PCI Passthrough of T1 cards (Mauricio Tavares) 7. Re: PCI Passthrough of T1 cards (Nick Khamis) 8. Fwd: AQuA Meter ? waveform analysis to get continous MOS scores for your network (Sevana Oy) -- Message: 1 Date: Mon, 8 Jul 2013 10:14:31 -0700 From: Justin Killen jkil...@allamericanasphalt.com Subject: [asterisk-users] analog phone digit delay To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 55b5d66c43b57f44bc89cb4650fd32f80118ffc2b...@mal.sg1.allamericanasphalt.com Content-Type: text/plain; charset=us-ascii I have an installation that has analog phones connected via T1 channel banks. I'm getting complaints from users that they will enter a partial number (eg 91213), then turn away to get the next few digits, and the system will start dialing before they have a chance to put in the rest of the dialing string. Is there a way to increase this delay? The only use these 4 dialing patterns: Internal 3 digit numbers 91 XXX XXX (for backwards compatibility) 9 XXX (also for compatibility) XXX I'm using the freepbx distro if that helps. Asterisk 11.2. Thanks, -Justin -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20130708/a09901d8/attachment-0001.htm -- Message: 2 Date: Mon, 08 Jul 2013 19:21:10 +0200 From: jg webaccou...@jgoettgens.de Subject: Re: [asterisk-users] analog phone digit delay To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 51daf506.5070...@jgoettgens.de Content-Type: text/plain; charset=UTF-8; format=flowed Have a look at the documentation of the channel bank. I guess some kind of overlap dialing is enabled, which is typically associated with a timeout value. chan_dahdi.conf also has entries like this. -- Message: 3 Date: Mon, 8 Jul 2013 10:45:52 -0700 From: Justin Killen jkil...@allamericanasphalt.com Subject: Re: [asterisk-users] analog phone digit delay To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 55b5d66c43b57f44bc89cb4650fd32f80118ffc2b...@mal.sg1.allamericanasphalt.com Content-Type: text/plain; charset=us-ascii 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. 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'. -Justin -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jg Sent: Monday, July 08, 2013 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay Have a look at the documentation of the channel bank. I guess some kind of overlap dialing is enabled, which is typically associated with a timeout value. chan_dahdi.conf also has entries like this. -- _ -- 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:
[asterisk-users] Global Variables
I have a question about global variables. Is it possible to somehow keep global variables unset via Dial Plan even Restarting asterisk? tks Eduardo -- _ -- 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
Re: [asterisk-users] Adjusting confbridge call quality
Hi, What codec do you use with yours subscribers? On 9 July 2013 23:45, Chris Gentle gent...@gmail.com wrote: Is there any way I can improve the audio quality in a confbridge in Asterisk 11? I've changed the internal_sample_rate setting to 44100 but that doesn't seem to make any difference. I would also think this would make my confbridge recordings 44100 but they all end up as 8000. Am I completely missing something? -- Chris -- _ -- 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
Re: [asterisk-users] Adjusting confbridge call quality
ulaw On Wed, Jul 10, 2013 at 7:40 AM, basteon bast...@gmail.com wrote: Hi, What codec do you use with yours subscribers? On 9 July 2013 23:45, Chris Gentle gent...@gmail.com wrote: Is there any way I can improve the audio quality in a confbridge in Asterisk 11? I've changed the internal_sample_rate setting to 44100 but that doesn't seem to make any difference. I would also think this would make my confbridge recordings 44100 but they all end up as 8000. Am I completely missing something? -- Chris -- _ -- 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 -- Chris -- _ -- 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
Re: [asterisk-users] Adjusting confbridge call quality
Chris Gentle wrote: Is there any way I can improve the audio quality in a confbridge in Asterisk 11? I've changed the internal_sample_rate setting to 44100 but that doesn't seem to make any difference. I would also think this would make my confbridge recordings 44100 but they all end up as 8000. Am I completely missing something? basteon wrote: What codec do you use with yours subscribers? Chris Gentle wrote: ulaw Chris, The sampling frequency for u-law is 8,000 Hz. You can't produce a recording with higher quality than the source, so you'd have to switch to a wideband codec to improve the conferences and recordings [1] [2]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats [2] http://www.slideshare.net/saghul/wideband-audio-conferencing-with-asterisk Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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
Re: [asterisk-users] Global Variables
Hi Eduardo Le 10/07/2013 14:30, Eduardo Leones a écrit : I have a question about global variables. Is it possible to somehow keep global variables unset via Dial Plan even Restarting asterisk? [...] From extensions.conf version 1.8 ; If clearglobalvars is set, global variables will be cleared ; and reparsed on an extensions reload, or Asterisk reload. ; ; If clearglobalvars is not set, then global variables will persist ; through reloads, and even if deleted from the extensions.conf or ; one if its included files, will remain set to the previous value. ; clearglobalvars=no Solution for restart is to use environement variables -- Daniel -- _ -- 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
Re: [asterisk-users] analog phone digit delay
Values for the timeouts just before the 'cannot complete as dialed, please try your call again': absolute: 0 digit: 5.000 response: 10.000 I've enabled DTMF logging to try to get a better log for interpretation. -Justin -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Monday, July 08, 2013 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] analog phone digit delay On Mon, 8 Jul 2013, Justin Killen wrote: I have an installation that has analog phones connected via T1 channel banks. I’m getting complaints from users that they will enter a partial number (eg 91213), then turn away to get the next few digits, and the system will start dialing before they have a chance to put in the rest of the dialing string. Is there a way to increase this delay? The only use these 4 dialing patterns: Will 'show function TIMEOUT' help? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
Re: [asterisk-users] Adjusting confbridge call quality
On Wed, Jul 10, 2013 at 9:16 AM, Matthew J. Roth mr...@imminc.com wrote: The sampling frequency for u-law is 8,000 Hz. You can't produce a recording with higher quality than the source, so you'd have to switch to a wideband codec to improve the conferences and recordings [1] [2]. OK, thanks for the info. I'm perfectly OK with 8,000 Hz except that I'm feeding the audio into a conference room from a microphone. chan_alsa actually is the first client to connect to the confbridge and then others can connect via SIP. For some reason, when the speaker says words with S's and F's, they almost sound distorted. Not quite static but you can tell the quality has been affected. May just be a side-effect of 8,000 Hz. Just wondered if there way some way to improve that. -- _ -- 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
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/1cannot-complete-as-dialedcheck-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:
Re: [asterisk-users] analog phone digit delay
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: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] 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/1cannot-complete-as-dialedcheck-number-dial-again,noanswer) in new stack [2013-07-10
[asterisk-users] queue moh
Hi All, Sorry if this has been covered already, but I don't tend to follow this list as close as I should these days. Problem is that if a call comes in to a queue without option 'r' specified - moh plays as expected. Now, when that call is answered, all is fine. Trouble comes when that person then puts the caller on-hold. No moh is heard by the caller (in fact, they get silence). If I use 'r' - then ringing is heard - but the queue's musiconhold/musicclass is ignored completely. When the caller is put on hold, they do hear moh but the default moh context is used - not the moh of the queue. What I need is for the queue's moh to be used when the caller is put on hold (and without using the 'r' feature). Is this possible? * 1.8.16.0 (tried on various flavours of 1.8). Queue static and realtime (same outcome). Cheers Andy -- If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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
[asterisk-users] autoanswer
Hello; To let the Phone answer automatically, this can be configured from asterisk (at the sip.conf for the phone)? Or it has to be from the IP Phone? Because, some phones does not support auto answer, also we do not need to do it for each Phone. Regards Bilal-- _ -- 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
Re: [asterisk-users] analog phone digit delay
On Mon, Jul 8, 2013 at 12:14 PM, Justin Killen jkil...@allamericanasphalt.com wrote: I have an installation that has analog phones connected via T1 channel banks. I’m getting complaints from users that they will enter a partial number (eg 91213), then turn away to get the next few digits, and the system will start dialing before they have a chance to put in the rest of the dialing string. Is there a way to increase this delay? The only use these 4 dialing patterns: ** ** Internal 3 digit numbers 91 XXX XXX (for backwards compatibility) 9 XXX (also for compatibility) XXX The simple switch in chan_dahdi has two hardcoded timeout times for more digits. 1) If the digits already dialed match an extension in the dialplan but could match another extension if more digits are dialed then chan_dahdi will wait 3 seconds for more digits to arrive. 2) If the digits already dialed do not match any extension in the dialplan but more digits could match an extension then chan_dahdi will wait 8 seconds for more digits. The shorter timeout is so the caller won't have to wait too long if the caller intends to call the shorter dialplan extension. You need to look at the extension patterns in your dialplan to see where you have ambiguity between extensions. Are you using the '.' wildcard? Richard -- _ -- 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
Re: [asterisk-users] queue moh
Hello Andy, Have you tried using SetMusicOnHold command before Queue command? BR, Ioan On Wed, Jul 10, 2013 at 7:55 PM, Andrew Thomas a...@datavox.co.uk wrote: Hi All, Sorry if this has been covered already, but I don't tend to follow this list as close as I should these days. Problem is that if a call comes in to a queue without option 'r' specified - moh plays as expected. Now, when that call is answered, all is fine. Trouble comes when that person then puts the caller on-hold. No moh is heard by the caller (in fact, they get silence). If I use 'r' - then ringing is heard - but the queue's musiconhold/musicclass is ignored completely. When the caller is put on hold, they do hear moh but the default moh context is used - not the moh of the queue. What I need is for the queue's moh to be used when the caller is put on hold (and without using the 'r' feature). Is this possible? * 1.8.16.0 (tried on various flavours of 1.8). Queue static and realtime (same outcome). Cheers Andy -- If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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
Re: [asterisk-users] analog phone digit delay
So then, by saying If the digits already dialed match an extension in the
dialplan...wait 3 seconds..., then we're saying that asterisk has found a
match, and the match is the bad-extension. Here is the bad-number context that
is included:
[bad-number]
include = bad-number-custom
exten = _X.,1,Noop(bad-number, timeouts: absolute: ${TIMEOUT(absolute)} digit:
${TIMEOUT(digit)} response: ${TIMEOUT(response)})
exten = _X.,n,ResetCDR()
exten = _X.,n,NoCDR()
exten = _X.,n,Progress
exten = _X.,n,Wait(1)
exten = _X.,n,Progress
exten =
_X.,n,Playback(silence/1cannot-complete-as-dialedcheck-number-dial-again,noanswer)
exten = _X.,n,Wait(1)
exten = _X.,n,Congestion(20)
exten = _X.,n,Hangup
So then, what you're saying then is that if I was to remove this include, there
would be no match in the dialplan and asterisk will wait for 8 seconds instead
of 3? The next question then is how to accomplish this without using the
wildcard (and how to change it in freepbx).
-Justin
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday, July 10, 2013 10:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
On Mon, Jul 8, 2013 at 12:14 PM, Justin Killen
jkil...@allamericanasphalt.commailto:jkil...@allamericanasphalt.com wrote:
I have an installation that has analog phones connected via T1 channel banks.
I'm getting complaints from users that they will enter a partial number (eg
91213), then turn away to get the next few digits, and the system will start
dialing before they have a chance to put in the rest of the dialing string. Is
there a way to increase this delay? The only use these 4 dialing patterns:
Internal 3 digit numbers
91 XXX XXX (for backwards compatibility)
9 XXX (also for compatibility)
XXX
The simple switch in chan_dahdi has two hardcoded timeout times for more digits.
1) If the digits already dialed match an extension in the dialplan but could
match another extension if more digits are dialed then chan_dahdi will wait 3
seconds for more digits to arrive.
2) If the digits already dialed do not match any extension in the dialplan but
more digits could match an extension then chan_dahdi will wait 8 seconds for
more digits.
The shorter timeout is so the caller won't have to wait too long if the caller
intends to call the shorter dialplan extension.
You need to look at the extension patterns in your dialplan to see where you
have ambiguity between extensions. Are you using the '.' wildcard?
Richard
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Re: [asterisk-users] Adjusting confbridge call quality
On 07/10/2013 06:46 PM, Chris Gentle wrote: [snip] and then others can connect via SIP. For some reason, when the speaker says words with S's and F's, they almost sound distorted. Not quite static but you can tell the quality has been affected. May just be a side-effect of 8,000 Hz. Just wondered if there way some way to improve that. The distorted S and F are prevented by a pop filter in front of the mic. Are you using a pop filter? Also if you are using a cheap mic, do yourself a favor and invest in a decent mic. It will make a world of difference. Regards, Patrick -- _ -- 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
Re: [asterisk-users] analog phone digit delay
From chan_dahdi.c, don't know if it applies to your situation or not.
/*! \brief Wait up to 16 seconds for first digit (FXO logic) */
static int firstdigittimeout = 16000;
/*! \brief How long to wait for following digits (FXO logic) */
static int gendigittimeout = 8000;
/*! \brief How long to wait for an extra digit, if there is an ambiguous match
*/
static int matchdigittimeout = 3000;
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July 10, 2013 3:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
So then, by saying If the digits already dialed match an extension in the
dialplan...wait 3 seconds..., then we're saying that asterisk has found a
match, and the match is the bad-extension. Here is the bad-number context that
is included:
[bad-number]
include = bad-number-custom
exten = _X.,1,Noop(bad-number, timeouts: absolute: ${TIMEOUT(absolute)} digit:
${TIMEOUT(digit)} response: ${TIMEOUT(response)})
exten = _X.,n,ResetCDR()
exten = _X.,n,NoCDR()
exten = _X.,n,Progress
exten = _X.,n,Wait(1)
exten = _X.,n,Progress
exten =
_X.,n,Playback(silence/1cannot-complete-as-dialedcheck-number-dial-again,noanswer)
exten = _X.,n,Wait(1)
exten = _X.,n,Congestion(20)
exten = _X.,n,Hangup
So then, what you're saying then is that if I was to remove this include, there
would be no match in the dialplan and asterisk will wait for 8 seconds instead
of 3? The next question then is how to accomplish this without using the
wildcard (and how to change it in freepbx).
-Justin
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday, July 10, 2013 10:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
On Mon, Jul 8, 2013 at 12:14 PM, Justin Killen jkil...@allamericanasphalt.com
wrote:
I have an installation that has analog phones connected via T1 channel banks.
I'm getting complaints from users that they will enter a partial number (eg
91213), then turn away to get the next few digits, and the system will start
dialing before they have a chance to put in the rest of the dialing string. Is
there a way to increase this delay? The only use these 4 dialing patterns:
Internal 3 digit numbers
91 XXX XXX (for backwards compatibility)
9 XXX (also for compatibility)
XXX
The simple switch in chan_dahdi has two hardcoded timeout times for more digits.
1) If the digits already dialed match an extension in the dialplan but could
match another extension if more digits are dialed then chan_dahdi will wait 3
seconds for more digits to arrive.
2) If the digits already dialed do not match any extension in the dialplan but
more digits could match an extension then chan_dahdi will wait 8 seconds for
more digits.
The shorter timeout is so the caller won't have to wait too long if the caller
intends to call the shorter dialplan extension.
You need to look at the extension patterns in your dialplan to see where you
have ambiguity between extensions. Are you using the '.' wildcard?
Richard
--
_
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Re: [asterisk-users] analog phone digit delay
On Wed, Jul 10, 2013 at 3:11 PM, Eric Wieling ewiel...@nyigc.com wrote: From chan_dahdi.c, don't know if it applies to your situation or not. /*! \brief Wait up to 16 seconds for first digit (FXO logic) */ static int firstdigittimeout = 16000; /*! \brief How long to wait for following digits (FXO logic) */ static int gendigittimeout = 8000; /*! \brief How long to wait for an extra digit, if there is an ambiguous match */ static int matchdigittimeout = 3000; Changing these values in chan_dahdi.c is unlikely to have any effect. You would need to change the equivalent versions in sig_analog.c instead. Richard -- _ -- 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
Re: [asterisk-users] analog phone digit delay
It seems likely that this is exactly what is happening. I'd rather not change
the code though, but rather fix the dialplan. I'm thinking using the 'i'
extension would work just the same - would there be a reason to use a wildcard
pattern match instead of i?
-Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, July 10, 2013 1:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
From chan_dahdi.c, don't know if it applies to your situation or not.
/*! \brief Wait up to 16 seconds for first digit (FXO logic) */
static int firstdigittimeout = 16000;
/*! \brief How long to wait for following digits (FXO logic) */
static int gendigittimeout = 8000;
/*! \brief How long to wait for an extra digit, if there is an ambiguous match
*/
static int matchdigittimeout = 3000;
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July 10, 2013 3:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
So then, by saying If the digits already dialed match an extension in the
dialplan...wait 3 seconds..., then we're saying that asterisk has found a
match, and the match is the bad-extension. Here is the bad-number context that
is included:
[bad-number]
include = bad-number-custom
exten = _X.,1,Noop(bad-number, timeouts: absolute: ${TIMEOUT(absolute)} digit:
${TIMEOUT(digit)} response: ${TIMEOUT(response)})
exten = _X.,n,ResetCDR()
exten = _X.,n,NoCDR()
exten = _X.,n,Progress
exten = _X.,n,Wait(1)
exten = _X.,n,Progress
exten =
_X.,n,Playback(silence/1cannot-complete-as-dialedcheck-number-dial-again,noanswer)
exten = _X.,n,Wait(1)
exten = _X.,n,Congestion(20)
exten = _X.,n,Hangup
So then, what you're saying then is that if I was to remove this include, there
would be no match in the dialplan and asterisk will wait for 8 seconds instead
of 3? The next question then is how to accomplish this without using the
wildcard (and how to change it in freepbx).
-Justin
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday, July 10, 2013 10:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
On Mon, Jul 8, 2013 at 12:14 PM, Justin Killen jkil...@allamericanasphalt.com
wrote:
I have an installation that has analog phones connected via T1 channel banks.
I'm getting complaints from users that they will enter a partial number (eg
91213), then turn away to get the next few digits, and the system will start
dialing before they have a chance to put in the rest of the dialing string. Is
there a way to increase this delay? The only use these 4 dialing patterns:
Internal 3 digit numbers
91 XXX XXX (for backwards compatibility)
9 XXX (also for compatibility)
XXX
The simple switch in chan_dahdi has two hardcoded timeout times for more digits.
1) If the digits already dialed match an extension in the dialplan but could
match another extension if more digits are dialed then chan_dahdi will wait 3
seconds for more digits to arrive.
2) If the digits already dialed do not match any extension in the dialplan but
more digits could match an extension then chan_dahdi will wait 8 seconds for
more digits.
The shorter timeout is so the caller won't have to wait too long if the caller
intends to call the shorter dialplan extension.
You need to look at the extension patterns in your dialplan to see where you
have ambiguity between extensions. Are you using the '.' wildcard?
Richard
--
_
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] analog phone digit delay
I has the same limitations as dialplan timeouts, you have to be in a
Background or WaitExten or similar for them to work.These items are
designed for IVRS.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July 10, 2013 4:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
It seems likely that this is exactly what is happening. I'd rather not change
the code though, but rather fix the dialplan. I'm thinking using the 'i'
extension would work just the same - would there be a reason to use a wildcard
pattern match instead of i?
-Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, July 10, 2013 1:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
From chan_dahdi.c, don't know if it applies to your situation or not.
/*! \brief Wait up to 16 seconds for first digit (FXO logic) */ static int
firstdigittimeout = 16000;
/*! \brief How long to wait for following digits (FXO logic) */ static int
gendigittimeout = 8000;
/*! \brief How long to wait for an extra digit, if there is an ambiguous match
*/ static int matchdigittimeout = 3000;
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July 10, 2013 3:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
So then, by saying If the digits already dialed match an extension in the
dialplan...wait 3 seconds..., then we're saying that asterisk has found a
match, and the match is the bad-extension. Here is the bad-number context that
is included:
[bad-number]
include = bad-number-custom
exten = _X.,1,Noop(bad-number, timeouts: absolute: ${TIMEOUT(absolute)} digit:
${TIMEOUT(digit)} response: ${TIMEOUT(response)})
exten = _X.,n,ResetCDR()
exten = _X.,n,NoCDR()
exten = _X.,n,Progress
exten = _X.,n,Wait(1)
exten = _X.,n,Progress
exten =
_X.,n,Playback(silence/1cannot-complete-as-dialedcheck-number-dial-again,noanswer)
exten = _X.,n,Wait(1)
exten = _X.,n,Congestion(20)
exten = _X.,n,Hangup
So then, what you're saying then is that if I was to remove this include, there
would be no match in the dialplan and asterisk will wait for 8 seconds instead
of 3? The next question then is how to accomplish this without using the
wildcard (and how to change it in freepbx).
-Justin
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday, July 10, 2013 10:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
On Mon, Jul 8, 2013 at 12:14 PM, Justin Killen jkil...@allamericanasphalt.com
wrote:
I have an installation that has analog phones connected via T1 channel banks.
I'm getting complaints from users that they will enter a partial number (eg
91213), then turn away to get the next few digits, and the system will start
dialing before they have a chance to put in the rest of the dialing string. Is
there a way to increase this delay? The only use these 4 dialing patterns:
Internal 3 digit numbers
91 XXX XXX (for backwards compatibility)
9 XXX (also for compatibility)
XXX
The simple switch in chan_dahdi has two hardcoded timeout times for more digits.
1) If the digits already dialed match an extension in the dialplan but could
match another extension if more digits are dialed then chan_dahdi will wait 3
seconds for more digits to arrive.
2) If the digits already dialed do not match any extension in the dialplan but
more digits could match an extension then chan_dahdi will wait 8 seconds for
more digits.
The shorter timeout is so the caller won't have to wait too long if the caller
intends to call the shorter dialplan extension.
You need to look at the extension patterns in your dialplan to see where you
have ambiguity between extensions. Are you using the '.' wildcard?
Richard
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Re: [asterisk-users] analog phone digit delay
Okay, so I is no good. Does anybody else have a work-around for this?
-Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, July 10, 2013 1:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
I has the same limitations as dialplan timeouts, you have to be in a
Background or WaitExten or similar for them to work.These items are
designed for IVRS.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July 10, 2013 4:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
It seems likely that this is exactly what is happening. I'd rather not change
the code though, but rather fix the dialplan. I'm thinking using the 'i'
extension would work just the same - would there be a reason to use a wildcard
pattern match instead of i?
-Justin
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, July 10, 2013 1:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
From chan_dahdi.c, don't know if it applies to your situation or not.
/*! \brief Wait up to 16 seconds for first digit (FXO logic) */ static int
firstdigittimeout = 16000;
/*! \brief How long to wait for following digits (FXO logic) */ static int
gendigittimeout = 8000;
/*! \brief How long to wait for an extra digit, if there is an ambiguous match
*/ static int matchdigittimeout = 3000;
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July 10, 2013 3:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
So then, by saying If the digits already dialed match an extension in the
dialplan...wait 3 seconds..., then we're saying that asterisk has found a
match, and the match is the bad-extension. Here is the bad-number context that
is included:
[bad-number]
include = bad-number-custom
exten = _X.,1,Noop(bad-number, timeouts: absolute: ${TIMEOUT(absolute)} digit:
${TIMEOUT(digit)} response: ${TIMEOUT(response)})
exten = _X.,n,ResetCDR()
exten = _X.,n,NoCDR()
exten = _X.,n,Progress
exten = _X.,n,Wait(1)
exten = _X.,n,Progress
exten =
_X.,n,Playback(silence/1cannot-complete-as-dialedcheck-number-dial-again,noanswer)
exten = _X.,n,Wait(1)
exten = _X.,n,Congestion(20)
exten = _X.,n,Hangup
So then, what you're saying then is that if I was to remove this include, there
would be no match in the dialplan and asterisk will wait for 8 seconds instead
of 3? The next question then is how to accomplish this without using the
wildcard (and how to change it in freepbx).
-Justin
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday, July 10, 2013 10:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] analog phone digit delay
On Mon, Jul 8, 2013 at 12:14 PM, Justin Killen jkil...@allamericanasphalt.com
wrote:
I have an installation that has analog phones connected via T1 channel banks.
I'm getting complaints from users that they will enter a partial number (eg
91213), then turn away to get the next few digits, and the system will start
dialing before they have a chance to put in the rest of the dialing string. Is
there a way to increase this delay? The only use these 4 dialing patterns:
Internal 3 digit numbers
91 XXX XXX (for backwards compatibility)
9 XXX (also for compatibility)
XXX
The simple switch in chan_dahdi has two hardcoded timeout times for more digits.
1) If the digits already dialed match an extension in the dialplan but could
match another extension if more digits are dialed then chan_dahdi will wait 3
seconds for more digits to arrive.
2) If the digits already dialed do not match any extension in the dialplan but
more digits could match an extension then chan_dahdi will wait 8 seconds for
more digits.
The shorter timeout is so the caller won't have to wait too long if the caller
intends to call the shorter dialplan extension.
You need to look at the extension patterns in your dialplan to see where you
have ambiguity between extensions. Are you using the '.' wildcard?
Richard
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[asterisk-users] Upgrade from 10.2.4 to 11.4.x, the Reader's Digest version?
Hi all, I'm contemplating an upgrade from 10.2.4 to 11.4.x. However, the 1.8.x to 10.4.x upgrade was painful; some of the modules had been renamed, if I recall correctly. So, is there a list of MAJOR changes and GOTCHA's between 10.x and 11.x? I'm hoping for something a little less granular than the release notes from 10.2.x to 11.4.x. I don't mind reading, but that is almost as long as War and Peace! Does such a document exist, or do I need to start reading.. TIA, Mike Diehl. -- _ -- 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
Re: [asterisk-users] Adjusting confbridge call quality
OK, thanks for the advice. No, there's no filter so I'll look into that. On Wed, Jul 10, 2013 at 3:02 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 07/10/2013 06:46 PM, Chris Gentle wrote: [snip] and then others can connect via SIP. For some reason, when the speaker says words with S's and F's, they almost sound distorted. Not quite static but you can tell the quality has been affected. May just be a side-effect of 8,000 Hz. Just wondered if there way some way to improve that. The distorted S and F are prevented by a pop filter in front of the mic. Are you using a pop filter? Also if you are using a cheap mic, do yourself a favor and invest in a decent mic. It will make a world of difference. Regards, Patrick -- _ -- 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 -- Chris -- _ -- 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
Re: [asterisk-users] autoanswer
On 07/10/2013 01:04 PM, bilal ghayyad wrote: Hello; To let the Phone answer automatically, this can be configured from asterisk (at the sip.conf for the phone)? Or it has to be from the IP Phone? Because, some phones does not support auto answer, also we do not need to do it for each Phone. Depends on the phone. Some phones you have to set auto answer on the phone itself, other phones can be told to auto answer via configuration file, and yet other phones can be told to auto answer if you send them the right SIP headers via the dialplan. -- _ -- 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
Re: [asterisk-users] Upgrade from 10.2.4 to 11.4.x, the Reader's Digest version?
On Wed, Jul 10, 2013 at 5:35 PM, Mike Diehl mdiehlena...@gmail.com wrote: Hi all, I'm contemplating an upgrade from 10.2.4 to 11.4.x. However, the 1.8.x to 10.4.x upgrade was painful; some of the modules had been renamed, if I recall correctly. So, is there a list of MAJOR changes and GOTCHA's between 10.x and 11.x? I'm hoping for something a little less granular than the release notes from 10.2.x to 11.4.x. I don't mind reading, but that is almost as long as War and Peace! Does such a document exist, or do I need to start reading.. Upgrade notes: https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 While the upgrade notes cover changes to configuration and module status, it is also a good idea to read through what is new: https://wiki.asterisk.org/wiki/display/AST/New+in+11 I wouldn't say it is War and Peace, but yes, there is some content in there. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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
