[asterisk-users] DTMF Tones occuring randomly
What codec are you currently using for voice? I have found that when nothing else works, playing with the gains on the Zap channel helped. Usually lowering them. I use rfc2833 for dtmf, alaw as codec. Yes, a lowering could be a idea, but the problem is logged on any kind of channels in my system, like zap, misdn, sip and iax. That is my problem :( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Tones occuring randomly
Hi Eric, i have replied but nobody seems to got a deeper knowledge of the problem. I have searched for talkoff, i found a lot of stuff, like check IRQs (checked, and good) and/or set relaxdtmf=no (it is set) or check the dtmf modes to be the same or or. But nothing of the things i found match to my problem except one thing i cant understand - there is an thread at digium with the advice to use the variable dtmfthreshold to set the level of dtmf detection, i cant find any variable like this. Do you know something where i can search ? I got this problem since 6 or 7 months and tried MANY solutions to get to my stable Asterisk, but i got no luck. What do you think about switching from rfc2833 to inband to solve this problem ? Thanks, Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Tones occuring randomly
Stefan Agethen wrote: Hi Eric, i have replied but nobody seems to got a deeper knowledge of the problem. I have searched for talkoff, i found a lot of stuff, like check IRQs (checked, and good) and/or set relaxdtmf=no (it is set) or check the dtmf modes to be the same or or. But nothing of the things i found match to my problem except one thing i cant understand - there is an thread at digium with the advice to use the variable dtmfthreshold to set the level of dtmf detection, i cant find any variable like this. Do you know something where i can search ? I got this problem since 6 or 7 months and tried MANY solutions to get to my stable Asterisk, but i got no luck. What do you think about switching from rfc2833 to inband to solve this problem ? What codec are you currently using for voice? I have found that when nothing else works, playing with the gains on the Zap channel helped. Usually lowering them. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Tones occuring randomly
What codec are you currently using for voice? I have found that when nothing else works, playing with the gains on the Zap channel helped. Usually lowering them. I use rfc2833 for dtmf, alaw as codec. Yes, a lowering could be a idea, but the problem is logged on any kind of channels in my system, like zap, misdn, sip and iax. That is my problem :( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Tones occuring randomly
I get the same thing using inband -- funny thing I am the only one who hears the random tones -- other party does not hear them and they are not recorded with the monitor app. on Wednesday 11/08/2006 Eric \ManxPower\ Wieling([EMAIL PROTECTED]) wrote Stefan Agethen wrote: Hi Eric, i have replied but nobody seems to got a deeper knowledge of the problem. I have searched for talkoff, i found a lot of stuff, like check IRQs (checked, and good) and/or set relaxdtmf=no (it is set) or check the dtmf modes to be the same or or. But nothing of the things i found match to my problem except one thing i cant understand - there is an thread at digium with the advice to use the variable dtmfthreshold to set the level of dtmf detection, i cant find any variable like this. Do you know something where i can search ? I got this problem since 6 or 7 months and tried MANY solutions to get to my stable Asterisk, but i got no luck. What do you think about switching from rfc2833 to inband to solve this problem ? What codec are you currently using for voice? I have found that when nothing else works, playing with the gains on the Zap channel helped. Usually lowering them. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Tones occuring randomly
Hi, I have asked this question months ago - i have toggled down all DTMF Recognizations in my Asterisk (no more features etc) and found more people which recognized the same problem, but i cant find any help for them and me. The Problem (short as possible) : In a randomly call in my business day some unit in my Asterisk System sends an randomly DTMF Tone, like A 0 or something that do something like # or *. In my case, the * let Asterisk hang up my call, i searched for help, but nobody knows what to do - so i disabled the hangup feature and so on, but the problem still exists :( I sets the hangup-function to : == Remapping feature Disconnect Call (disconnect) to sequence '*0' My System is a : Asus with an AMD Athlon XP 3000+ with 512MB of RAM, 1 Wildcard TDM40B, 2 HFC ISDN PCI Cards from Acer (128k Surf). Installed is : Debian 3.1 with unstable packages to get Kernel 2.6.15-1 (AMD Kernel) (in earlier days my ISDN Driver, mISDN only works with Kernel 2.6.12 or higher, Debian is 2.6.8, so...) The needed Packages for Asterisk are installed (My Installation Step-by-Step in german is here : http://www.ip-phone-forum.de/showpost.php?p=657963postcount=7) Zaptel 1.2.9 Asterisk 1.2.12.1 mISDN in 0.3.0 RC 23 I have changed mpeg123 against madplay. The Problem exists since a half year or more, i like to say it in another way : i have RECOGNIZED the problem since a half year, i have done many updates of all packages and a clean install to merge this prob, no luck, it still exists. The facts i know about it : During such a * DTMF Shooting the logfiles recognized this (see the channel types!) : -- NOTICES -- Nov 6 09:53:26 WARNING[22637] res_features.c: Bridge failed on channels mISDN/1-1 and Zap/1-1 Nov 6 10:05:28 WARNING[22902] res_features.c: Bridge failed on channels SIP/40-0815e778 and SIP/pbx1-08281bc8 Nov 6 10:15:38 WARNING[23393] res_features.c: Bridge failed on channels SIP/40-0826c530 and IAX2/pbx1-1 DTMF Tone Log : Nov 6 05:00:33 DTMF[18215] channel.c: SIP/50-0824f1e0 : A Nov 6 08:44:05 DTMF[21660] channel.c: Zap/1-1 : A Nov 6 09:43:00 DTMF[22520] channel.c: SIP/pbx1-08274fb8 : 0 Nov 6 09:53:26 DTMF[22637] channel.c: Zap/1-1 : * Nov 6 10:05:28 DTMF[22902] channel.c: SIP/pbx1-08281bc8 : * Nov 6 10:14:42 DTMF[23288] channel.c: mISDN/2-1 : 8 Nov 6 10:16:11 DTMF[23426] channel.c: SIP/pbx1-08274690 : * Nov 6 10:17:45 DTMF[23288] channel.c: Zap/1-1 : A Nov 6 10:32:54 DTMF[23545] channel.c: Zap/1-1 : D Nov 6 10:35:58 DTMF[23792] channel.c: SIP/pbx1-08273ef8 : * -- ASTERISK SIP DEBUG (one case) -- Nov 6 10:35:54 DEBUG[23792] channel.c: Got DTMF on channel (SIP/40-0825b3c8) Nov 6 10:35:54 DEBUG[23792] channel.c: Bridge stops bridging channels SIP/40-0825b3c8 and SIP/pbx1-08273ef8 Nov 6 10:35:54 DEBUG[23792] res_features.c: Feature interpret: chan=SIP/40-0825b3c8, peer=SIP/pbx1-08273ef8, sense=1, features=18 Nov 6 10:35:54 DEBUG[23792] res_features.c: Set time limit to 500 Nov 6 10:35:55 DEBUG[23792] channel.c: Nobody there, continuing... Nov 6 10:35:58 DEBUG[23792] channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/40-0825b3c8, c1=SIP/pbx1-08273ef8, flags: No,Yes,No,No Nov 6 10:35:58 DEBUG[23792] channel.c: Bridge stops bridging channels SIP/40-0825b3c8 and SIP/pbx1-08273ef8 Nov 6 10:35:58 DEBUG[23792] res_features.c: Timed out for feature! Nov 6 10:35:58 DEBUG[23792] channel.c: Hanging up channel 'SIP/pbx1-08273ef8' Nov 6 10:35:58 DEBUG[23792] chan_sip.c: Hangup call SIP/pbx1-08273ef8, SIP callid [EMAIL PROTECTED]) Nov 6 10:35:58 DEBUG[23792] chan_sip.c: update_call_counter(02088480499) - decrement call limit counter Nov 6 10:35:58 DEBUG[23792] app_dial.c: Exiting with DIALSTATUS=ANSWER. Nov 6 10:35:58 DEBUG[23792] pbx.c: Spawn extension (voip_wahl,_X.,6) exited non-zero on 'SIP/40-0825b3c8' As you can see in the DTMF Log - there are many Digits send, but they dont scare me, but the * are disconnecting my calls - thats a problem for me and my business.. I HOPE !!! you can help me, Best wishes, Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Tones occuring randomly
Hi, I have asked this question months ago - i have toggled down all DTMF Recognizations in my Asterisk (no more features etc) and found more people which recognized the same problem, but i cant find any help for them and me. The Problem (short as possible) : In a randomly call in my business day some unit in my Asterisk System sends an randomly DTMF Tone, like A 0 or something that do something like # or *. In my case, the * let Asterisk hang up my call, i searched for help, but nobody knows what to do - so i disabled the hangup feature and so on, but the problem still exists :( I sets the hangup-function to : == Remapping feature Disconnect Call (disconnect) to sequence '*0' My System is a : Asus with an AMD Athlon XP 3000+ with 512MB of RAM, 1 Wildcard TDM40B, 2 HFC ISDN PCI Cards from Acer (128k Surf). Installed is : Debian 3.1 with unstable packages to get Kernel 2.6.15-1 (AMD Kernel) (in earlier days my ISDN Driver, mISDN only works with Kernel 2.6.12 or higher, Debian is 2.6.8, so...) The needed Packages for Asterisk are installed (My Installation Step-by-Step in german is here : http://www.ip-phone-forum.de/showpost.php?p=657963postcount=7) Zaptel 1.2.9 Asterisk 1.2.12.1 mISDN in 0.3.0 RC 23 I have changed mpeg123 against madplay. The Problem exists since a half year or more, i like to say it in another way : i have RECOGNIZED the problem since a half year, i have done many updates of all packages and a clean install to merge this prob, no luck, it still exists. The facts i know about it : During such a * DTMF Shooting the logfiles recognized this (see the channel types!) : -- NOTICES -- Nov 6 09:53:26 WARNING[22637] res_features.c: Bridge failed on channels mISDN/1-1 and Zap/1-1 Nov 6 10:05:28 WARNING[22902] res_features.c: Bridge failed on channels SIP/40-0815e778 and SIP/pbx1-08281bc8 Nov 6 10:15:38 WARNING[23393] res_features.c: Bridge failed on channels SIP/40-0826c530 and IAX2/pbx1-1 DTMF Tone Log : Nov 6 05:00:33 DTMF[18215] channel.c: SIP/50-0824f1e0 : A Nov 6 08:44:05 DTMF[21660] channel.c: Zap/1-1 : A Nov 6 09:43:00 DTMF[22520] channel.c: SIP/pbx1-08274fb8 : 0 Nov 6 09:53:26 DTMF[22637] channel.c: Zap/1-1 : * Nov 6 10:05:28 DTMF[22902] channel.c: SIP/pbx1-08281bc8 : * Nov 6 10:14:42 DTMF[23288] channel.c: mISDN/2-1 : 8 Nov 6 10:16:11 DTMF[23426] channel.c: SIP/pbx1-08274690 : * Nov 6 10:17:45 DTMF[23288] channel.c: Zap/1-1 : A Nov 6 10:32:54 DTMF[23545] channel.c: Zap/1-1 : D Nov 6 10:35:58 DTMF[23792] channel.c: SIP/pbx1-08273ef8 : * -- ASTERISK SIP DEBUG (one case) -- Nov 6 10:35:54 DEBUG[23792] channel.c: Got DTMF on channel (SIP/40-0825b3c8) Nov 6 10:35:54 DEBUG[23792] channel.c: Bridge stops bridging channels SIP/40-0825b3c8 and SIP/pbx1-08273ef8 Nov 6 10:35:54 DEBUG[23792] res_features.c: Feature interpret: chan=SIP/40-0825b3c8, peer=SIP/pbx1-08273ef8, sense=1, features=18 Nov 6 10:35:54 DEBUG[23792] res_features.c: Set time limit to 500 Nov 6 10:35:55 DEBUG[23792] channel.c: Nobody there, continuing... Nov 6 10:35:58 DEBUG[23792] channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/40-0825b3c8, c1=SIP/pbx1-08273ef8, flags: No,Yes,No,No Nov 6 10:35:58 DEBUG[23792] channel.c: Bridge stops bridging channels SIP/40-0825b3c8 and SIP/pbx1-08273ef8 Nov 6 10:35:58 DEBUG[23792] res_features.c: Timed out for feature! Nov 6 10:35:58 DEBUG[23792] channel.c: Hanging up channel 'SIP/pbx1-08273ef8' Nov 6 10:35:58 DEBUG[23792] chan_sip.c: Hangup call SIP/pbx1-08273ef8, SIP callid [EMAIL PROTECTED]) Nov 6 10:35:58 DEBUG[23792] chan_sip.c: update_call_counter(02088480499) - decrement call limit counter Nov 6 10:35:58 DEBUG[23792] app_dial.c: Exiting with DIALSTATUS=ANSWER. Nov 6 10:35:58 DEBUG[23792] pbx.c: Spawn extension (voip_wahl,_X.,6) exited non-zero on 'SIP/40-0825b3c8' As you can see in the DTMF Log - there are many Digits send, but they dont scare me, but the * are disconnecting my calls - thats a problem for me and my business.. I HOPE !!! you can help me, Best wishes, Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Tones occuring randomly
Stefan Agethen wrote: Hi, I have asked this question months ago - i have toggled down all DTMF Recognizations in my Asterisk (no more features etc) and found more people which recognized the same problem, but i cant find any help for them and me. The problem is called Talk Off (or maybe Talkoff). Search the archives for that. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
