similar to: Missing 31 DTMF tones over ZAP

I see there are still some floating point operations left in the codec init(ialization) code. Changing that code to fixed point is not only difficult (due to the trigonometric functions etc) but may also degrade the precision. Here is an idea whereby we can easily precompute (record) all those values on a powerful processor and then use (replay) them on an embedded processor / DSP. The only

Hello Adam, Thank you so much for taking the time to write to me. I can understand your concerns; let me see if I can address them. >Matt, > >Sourceforge.net is exclusively for hosting software whose licensing >terms meet the OSI's definition of Open Source: > > http://opensource.org/docs/definition.php > >Your licensing terms include the following, which is

cdr_mysql doesn't set the userfield when it's set inside a macro called from a feature (1.4.25, addons 1.4.8). I have a feature code: autorecord => *1,self,Macro,apprecord The apprecord macro looks like: [macro-apprecord] exten => s,1,Playback(beep) exten =>

Hello. I notice that when a call that is recorded with MixMonitor is transfered to another co-worker, the recording ends. exten => 409,n,Macro(SDstartrecording,external,${DID}) the incoming call then goes to a queue... [macro-startrecording] ; ARG1 = incoming DID or CALLERID(name) ; ARG2 = outgoing dialnumber ... exten => s,n,MixMonitor(/var/ftp/${NR}/${recordfile},b,chown -R

All; I wrote a PERL AGI script that prompts a caller to leave a message using print "RECORD FILE $recordfile wav # 60000 BEEP s=3\n"; When the caller is done, they need to press the # key. The message is then delivered. However, the message is not delivered if the caller simply hangs up when finished. If the user hangs up, the script ends right then. How do I keep on processing the

I have a plot where the values of the y axis go from a positive number to a negative number and I want the x axis to intercept at zero rather than at the bottom of the y axis, regardless of its value. Can anyone help me to do this? Thanks in advance Vivien Vivien Kent MSc Oxon PhD candidate Evolutionary Anthropology Research Group Department of Anthropology Durham University Dawson

I'm using asterisk 1.2.14 When asterisk stores voicemail messages in /var/spool/asterisk/voicemail/default/EXTENSION/INBOX files are created with: -rwx------ 1 asterisk web-aster 6690 Apr 17 16:08 msg0002.WAV -rwx------ 1 asterisk web-aster 6732 Apr 17 16:08 msg0002.gsm -rw------- 1 asterisk web-aster 274 Apr 17 16:08 msg0002.txt -rwx------ 1 asterisk web-aster 65324 Apr 17 16:08

Hello list, I had to boot a Dell Diagnostic Disk today, and tried doing this via memdisk - which was when I noticed memdisk isn't working for me at all, neither from pxelinux nor syslinux. After trying around with 'raw', 'bigraw' and the like I noticed it doesn't even do anything when just using: LABEL diag KERNEL memdisk .. when I try that: boot: diag nothing

Hey folks, I have a Digium TE205P working as a man in the middle: PRI line -------- Asterisk/TE205P -------- PBX The PBX is a Panasonic KX - TVP 100. Everything is working great except for one little issue. Asterisk isn't hanging up the PRI B channel when the PBX channel is hung up. I don't want to overload you with information but please ask if more is needed. I suspect I'm

Currently I am having 2 issues with my Avaya 4602 phone: First, the phone registers with my Asterisk server, but when I start dialing I get a busy signal after 4 digits. I specified in the dialplan on the phone to expect 10 digits and that solved that problem, but I still immediately get a busy after the 10th digit. The phone never sends a dial command to asterisk. Second, asterisk is

Hello, I know there's a problem with Asterisk at the moment in that while it's easy for one caller to dial another (using the dial command), it's tricky to connect two calls that are already in progress. I've been using MeetMe to achieve this (with each caller's call being directed to a dynamically created conference room programatically), and this is working - kind of -

https://bugzilla.samba.org/show_bug.cgi?id=5459 Summary: Large amount of files makes checksum count negative Product: rsync Version: 3.0.2 Platform: x64 OS/Version: Linux Status: NEW Severity: major Priority: P3 Component: core AssignedTo: wayned@samba.org ReportedBy:

I'm trying to play a wav-file on a channel. This is what I see in the asterisk debug console AGI Rx << STREAM FILE "test.wav" "12345" [Aug 20 16:10:19] WARNING[25219]: file.c:602 ast_openstream_full: File test.wav does not exist in any format So it doesn't find the file, or it's in a wrong format? I can listen to it with windows media player... it's a

The company I work for has deployed an Avaya IP phone system. They have deployed the Avaya 4602 and 4620 IP telephones. They might be sending me one of these phones for use in my home office. Question: Can I make this IP telephone register and work with my Asterisk server? I don't know if it is a SIP phone? I searched thru the Avaya site, but can't find whether it's a SIP phone or

Hi, I am installing xen first time. Install in suse linex 9.3 & red hat se 4.3 but i could not start xen domain. I have installation guide tar files & rpms of xen 3.0. but still i am not able co to configure xen 3.0. can some body help me how to configure xen. Awaiting for reply. __________________________________________________________ Yahoo! India Answers: Share what you

Anyone have an H.323 trunk tied between their Avaya and Asterisk box that works? I am having some issues trying to get the two systems to connect. I am using the ooh323 channel to try to make the connection between the two system. I have all my configs if anyone would like to look over them. If I do a trace on Avaya I get a denial event 1191: Network Failure. Thanks! -------------- next part

Hi ! I am trying to connect Asterisk with Avaya Definity. I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything Example Asterisk ---> Avaya --

So far I have not found a way that I can register the Avaya phone with Asterisk. From what I have found so far is that Avaya phone needs the Avaya Media Server and Avaya Gateway. Looking at the h.323.conf (in Asterisk) and the file 46xxsettings.txt (avaya file located in tftpboot) there are no settings to make the phone initialize. I have sent an email to the Asterisk Users Mailing List to see

I would like to connect a Legacy PBX (Avaya IP Office 406) to an Asterisk Server. The Avaya has 3 CO Ports available. I thought buying a TDM30B card with 3 FXS ports to connect the * to the Avaya CO Ports. Is this the right approach? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions.... Daniel Kocher

Hi all, I've been pulling my hair out for two days over this problem... I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem