Re: [Asterisk-Users] i4l-modem dtmf detection
server:/usr/src/linux/drivers/isdn# patch -p0 < ../../../isdn-kernel-dtmf-dsp-patch.diff patching file isdn_tty.c patch: malformed patch at line 9: (info->emu.vpar[1])) what can be this?? Matthew Enger wrote: And a working patch for linux kernel. On Fri, 2003-11-07 at 09:30, Matthew Enger wrote: The correct URL is http://www.marko.net/asterisk/archives/0301/0849.html for those who want it. Regards, Matthew Enger [EMAIL PROTECTED] On Thu, 2003-11-06 at 09:34, Matthew Enger wrote: Hello, You need to apply two patches: 1) Turns off DTMF detection in the linux kernel (i4l side) 2) Enables DTMF detection on the chan_modem driver. You can find more information at http://www.marko.net/asterisk/archives/30301/0849.html The kernel is detecting DTMF tones from your voice :) I applied this yesterday and it is working great (kernel 2.4.22 and latest asterisk cvs) Regards, Matthew Enger [EMAIL PROTECTED] On Thu, 2003-11-06 at 05:34, Tomaz Izanc wrote: hello! I have active call from i4l modem to ZAP (FXS).When someone on i4l (telco side) speaks i hear DTMF tones on other side (ZAP). How to turn off DTMF detection on modem-i4l side ? Is it possible to do that ?? status of active channels: server*CLI> show channel Modem[i4l]/ttyi0 -- General -- Name: Modem[i4l]/ttyI0 Type: Modem UniqueID: 1068056585.53 Caller ID: 5 DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormat: 64 WriteFormat: 64 ReadFormat: 64 1st File Descriptor: 8 Frames in: 10914 Frames out: 7514 Time to Hangup: 0 -- PBX -- Context: remote Extension: 0346546777 Priority: 2 Call Group: 0 Pickup Group: 0 Application: Dial Data: Zap/1/0346546777w||r Stack: 0 Blocking in: ast_waitfor_nandfds - server*CLI> show channel Zap/1-1 -- General -- Name: Zap/1-1 Type: Zap UniqueID: 1068056588.54 Caller ID: 5 DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormat: 68 WriteFormat: 64 ReadFormat: 64 1st File Descriptor: 18 Frames in: 5536 Frames out: 6378 Time to Hangup: 0 -- PBX -- Context: nme Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: Bridged Call Data: Modem[i4l]/ttyI0 Stack: -1 Blocking in: ast_waitfor_nandfds - server*CLI> zap show channel 1 Channel: 1> File Descriptor: 18 Span: 1 Extension: Context: nmt Caller ID string: Destroy: 0 Signalling Type: FXS Kewlstart Owner: Zap/1-1 Real: Zap/1-1 (Linear) Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: yes Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently ON Actual Confinfo: Num/0, Mode/0x Actual Confmute: No tnx. Tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- /root/isdn_tty.c 2003-11-05 18:02:49.0 +1100 +++ isdn_tty.c 2003-11-05 18:08:46.0 +1100 @@ -133,9 +133,9 @@ if (info->online) { r = 0; #ifdef CONFIG_ISDN_AUDIO -isdn_audio_eval_dtmf(info); -if ((info->vonline & 1) && (info->emu.vpar[1])) - isdn_audio_eval_silence(info); +//isdn_audio_eval_dtmf(info); +//if ((info->vonline & 1) && (info->emu.vpar[1])) +// isdn_audio_eval_silence(info); #endif if ((tty = info->tty)) { if (info->mcr & UART_MCR_RTS) { @@ -190,10 +190,10 @@ #ifdef CONFIG_ISDN_AUDIO ifmt = 1; - if ((info->vonline) && (!info->emu.vpar[4])) - isdn_audio_calc_dtmf(info, skb->data, skb->len, ifmt); - if ((info->vonline & 1) && (info->emu.vpar[1])) - isdn_audio_calc_silence(info, skb->data, skb->len, ifmt); + //if ((info->vonline) && (!info->emu.vpar[4])) + // isdn_audio_calc_dtmf(info, skb->data, skb->len, ifmt); + //if ((info->vonline & 1) && (info->emu.vpar[1])) + // isdn_audio_calc_silence(info, skb->data, skb->len, ifmt); #endif if ((info->online < 2) #ifdef CONFIG_ISDN_AUDIO ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] i4l-modem dtmf detection
hello! I have active call from i4l modem to ZAP (FXS).When someone on i4l (telco side) speaks i hear DTMF tones on other side (ZAP). How to turn off DTMF detection on modem-i4l side ? Is it possible to do that ?? status of active channels: server*CLI> show channel Modem[i4l]/ttyi0 -- General -- Name: Modem[i4l]/ttyI0 Type: Modem UniqueID: 1068056585.53 Caller ID: 5 DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormat: 64 WriteFormat: 64 ReadFormat: 64 1st File Descriptor: 8 Frames in: 10914 Frames out: 7514 Time to Hangup: 0 -- PBX -- Context: remote Extension: 0346546777 Priority: 2 Call Group: 0 Pickup Group: 0 Application: Dial Data: Zap/1/0346546777w||r Stack: 0 Blocking in: ast_waitfor_nandfds - server*CLI> show channel Zap/1-1 -- General -- Name: Zap/1-1 Type: Zap UniqueID: 1068056588.54 Caller ID: 5 DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormat: 68 WriteFormat: 64 ReadFormat: 64 1st File Descriptor: 18 Frames in: 5536 Frames out: 6378 Time to Hangup: 0 -- PBX -- Context: nme Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: Bridged Call Data: Modem[i4l]/ttyI0 Stack: -1 Blocking in: ast_waitfor_nandfds - server*CLI> zap show channel 1 Channel: 1> File Descriptor: 18 Span: 1 Extension: Context: nmt Caller ID string: Destroy: 0 Signalling Type: FXS Kewlstart Owner: Zap/1-1 Real: Zap/1-1 (Linear) Callwait: Threeway: Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: yes Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently ON Actual Confinfo: Num/0, Mode/0x Actual Confmute: No tnx. Tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dtmf detection on modem-ISDN
hello! How to turn off DTMF detection on modem (isdn) on active channel (when the channel is open )? Problem is that dtmf is detected when someone on (ISDN) telco side speak then dtmf tones are send to from asterisk to internal line (x100p) . tnx. Tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] grandstream-budge tone message button
hi! Is it possible to get working message button on grandstream-budge tone phone ? For call to VM and also to signaling messages in VM ? or at least any of this two. tnx, Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incomplete address response SIP 484
hello! Is maybe anyway that asterisk supports this "incomplete address response" SIP 484 (early dial) ? I think it would be nice feature for dial with hard sip phones .. now must wait with all sip phones ~ 4 seconds or press # , but when your "dial frequency" is high is this somehow disturbing. thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip dial command 484
hi! Is asterisk support SIP early dial response 484 ? how to setup if .. tnx,tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] siemens optipoint 400 SIP
hi! anyone try siemens optipoint 400 economy SIP phone with * ? -- http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialogic DIALOG/4
how you mean "board is not full duplex"? ofcourse is full duplex it has been used for iptelephony "long" time ago. have you try this board in linux? you have drivers? i have two cards ;) Thomas Benjamin Miller wrote: This board is not full duplex and will not work with *. I have 3 of them :-( Buy some Digium boards. They will serve you much better and are easier. Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Sent: Wednesday, May 28, 2003 3:22 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] dialogic DIALOG/4 Good question. This board works well with * ? Can I get the caller-id(ANI) using this board? Isamar On Wed, 28 May 2003, Tomaz Izanc wrote: hi .. anyone using dialogic isa board DIALOG / 4 ? Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialogic DIALOG/4
hi .. anyone using dialogic isa board DIALOG / 4 ? Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] callerid
"In general you can match callerID with the /, but if you don't put anything after the /, then the rule matches "no caller*ID", and if no slash is there at all, it matches "any callerid". " Ok.My question is -> how to match callerid from 001... ? and if don't know how many numbers ? exten => s/0_,Answer don't work- anything else ? tnx Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users