similar to: How to change outgoing DTMF frequencies on zaptel?

Hi, After some testing I've found out that my client's hardware recognizes DTMF only if digits are sent 50ms apart with 50ms of tone duration. This was tested using a test device which generates DTMF. Now asterisk doesn't do it by default because digits going out from Asterisk are not being recognized. Using command sendDTMF, I can control inter-digit duration, and using

Hello, I noticed that changes to realtime sip peers are not applied until a 'reload'. A 'sip reload' does not make any changes to realtime sip peers. When changing for instance the mailbox-parameter in the realtime sip_buddies table, the change is not applied with a 'sip reload'. For every change there is a complete 'reload' necessary. Why does a 'sip

Hi I'm sorry for the my trivial quest. I Have asterisk 1.4 with TDM 400 with FXO and FXS, and works fine from several months. Now I want to connect a device to TDMFXS that want a ring frequecy of 25 hz to activate: i am italian, and usually the ring freq is 20 hz. The other time (I have used that device several times with other asterisk installation) I have modified /etc//modprobe.conf and

Hello, we have an asterisk (1.4.21.2) with ZAP and mISDN channels, the mISDN are 2 incoming GSM Modems, each with 2 simcards. No, the mISDN line one and two are fine, but when I get a call on line 3 something with the time is wrong. Timeconditions fall through to off-hours even if the time of the call is clearly inside business hours, here a log excerpt: [Aug 26 11:04:36] VERBOSE[3112]

Anyone else get this while compiling zaptel? I'm guessing I have to modify my kernel. Neato. :( Does that mean that the zaptel module (I'm really after ztdummy), or this xpp_zap thing won't be usable...? Not that I have zaptel hardware, but it seems Asterisk won't compile itself without zaptel being installed. CC [M] /root/software/zaptel-1.2.6/xpp/xpp_zap.o

Hi, I can't figure out what syntax to use with the Dial() "M" parameter for the AEL parser to interpret properly. Creating an AEL macro named "macro-screen()" partly works as a hack, but it must not turn into a gosub properly, so I get warnings about the "return;". Dial(...,tgM(&screen)) with the ael macro named "screen" does not work

Hi Jean-Marc, You make that sound like its just a matter of meeting some arbitrary spec. Let's be more specific..... If you use narrow band voice down to deep bass frequencies: - 16 bit linear audio sounds good - alaw or ulaw sounds muddy - low bit rate codecs, like speex or G.729, sound awful. I assume QinBin only listened to some uncompressed audio in his evaluation.

Hi, some days ago we discused here the need for an analyzer for the 1000 Hz tone, as opposite application to Milliwatt. Here it is: Mwanalyze http://planinternet.net/download/voip/asterisk/app_mwanalyze.c It performs a Fourier analysis for a fixed frequency and tells the amplitude. The frequency is not limited to 1000 Hz, but can be passed as argument. The periode duration must be a mulitple

Hi everybody, I am having DTMF detection problem on DISA with my callback system. For many users, it keeps playing the dialtone even after they have input their number. I have trunk setup to both g729 and ulaw. What could be the reason for this problem. Some users have to dial a few times before the system can recognize their dialed number. -- Zeeshan A Zakaria -------------- next part

Hello Everyone, We are running asterisk 1.2.x version in production environment since last 5 year and we have no issue at all, But now time to upgrade. and i heard about 1.8 which has introduce many features. I am wondering should I use asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version? I would like if you suggest me which version would be good for production since

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I wrote a script that I anticipating seeing a spike at 10Hz with the function 10* sin(2*pi*10*t). I can't figure out why my plots do not show spikes at the frequencies I expect. Am I doing something wrong or is my expectations wrong? require(stats) layout(matrix(c(1,2,3), 3, 1, byrow = TRUE)) #SETUP n <- 256 #nextn(1001) gives next power 2 F <- 100 #Hz -50 to 50 Hz dt

Hi all. I just upgraded my server from 4.7-Stable to 5.0-Release by a full install. Now I hit a problem with the date.. the time goes much faster than normal.. maybe twice as normal... this morning I found the following kernel log in /var/log/messages May 26 17:23:58 jupiter kernel: Timecounter "i8254" frequency 1193182 Hz May 26 17:23:58 jupiter kernel: Timecounter "TSC"

Dear list, Could you please tell me the values of the minimum and maximum cutoff frequencies for each coding version of the 44.1 kHz sampled data? For instance, are the values fmin=100 Hz and fmax=12 kHz valid? Thank you very much in advance. Kind regards, ? Fernando A. Marengo Rodriguez, PhD Post-doctoral fellow on Acoustics and Beamforming -- Laboratory of Noise and Vibration (LVA) Federal

Ok here is a real geek question, I building my own linux kernel for my asterisk system and came across the kernel setting for the timer frequency. I have one of 3 hardcode choices 100Hz, 250 Hz and 1000Hz. From what I understand the default Freq was changed from 100Hz in kernel 2.4 to 1000Hz (1KHz) in kernel 2.6. Timing is a BIG issue in asterisk with all the TDM and zap channel stuff.

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over SIP trunk from which calls get routed to third server (C) (1.6.0.9) via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP

Hi, Is it possible to create a clear zaptel channel which doesn't require to be picked up? The requirement of my client is to open a clear channel to a recorder which starts recording certain message. Currently the channel which is created by zaptel requires the other end to answer the call, and the recorded can't answer, so the channel get hung up after a certain number of rings. Zaptel

Hi, I am using zap channels, and by using sendDTFM application, I can control the duration between two DTMF digits, but I can't find a way to control the duration of the digits themself. Did search on the Internet and found out that I can change it in the asterisk source files and recompile asterisk. Wiki also says that it can be controlled using toneduration option in zapata.conf, but it

Hi, But in fact, it really affects the voice quality. One of my tester says, "Is your mouth far way from the mic?" Could you explain why we should cut 200hz below? >The notch filter is specifically designed to cut below 200 Hz when >working in narrowband. In wideband, the cutoff is more around 50 Hz. The >reason is that in narrowband operation (irrespective of the

Hi, My solaris storage hangs. I login to the console and there is messages[1] display on the console. I can''t login into the console and seems the IO is totally blocked. The system is solaris 10u8 on Dell R710 with disk array Dell MD3000. 2 HBA cable connect the server and MD3000. The symptom is random. It is very appreciated if any one can help me out. Regards, Ding [1] Aug 16