search for: dplatt

> > Different brand/model, but similar as they are both el cheapo, > entry-level headsets. I tried using them on a laptop, and I get > marginally better microphone output, even with its volume cranked all > the way up + automatic gain control enabled. > > I guess those on-board soundcards by Realtek aren't as good as a > quality microphones. I'll get a USB headset

...users] recording in mp3 </div><div> </div>no need. mixmonitor has a argument that is a script ran just as the recording is finished. we use this to move the file from ramfs to final destination. you can use it to use sox and convert it... On 2 July 2014 18:54, Dave Platt <dplatt at radagast.org> wrote: > Problem with this is client needs to listen to the call recordings and my interface will only display .wav or .mp3 so they will moan if they have to wait until the next day for today's recordings If you're up to writing a bit of shell script, and are runnin...

I've run into a odd issue where inbound calls to the SIP client work fine, but outbound from the SIP client do not. The path between the client and the server is as below. N900 SIP client <-- OpenVPN --> Asterisk The version of Asterisk in question is 1.6.0.18. Any suggestions? -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type:

> Now I have the problem for my cellphone... I need to register from almost any > IP (at least in Europe), so I can't restrict it. > Well, the password is NOT simple and random. > > Now, I tried to register the user of my cellphone using a PC, as my cellphone > was already registered. > And Asterisk accepted this registration... :( Were you trying to register the PC

> Problem with this is client needs to listen to the call recordings and my interface will only display .wav or .mp3 so they will moan if they have to wait until the next day for today's recordings If you're up to writing a bit of shell script, and are running on Linux, you could automate the conversion process so that it happens as soon as the recording is completed. Look at the

>> See if it plays back properly. > > Running aplay as asterisk user seems to be no problem: > > asterisk at puppy$ aplay /usr/share/sounds/alsa/Front_Center.wav > Playing: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit > Little Endian, Rate: 48000 Hz, mono > asterisk at puppy:~$ aplay -Dpulse /usr/share/sounds/alsa/Front_Center.wav >

OK, I have windoze ME installed on my system and have been trying to run IE 6.0 with wine release 20010824. Trying to start iexplore.exe goes along until: Unhandled exception: page fault on read access to 0xdeadbeef in 32-bit code (0xdeadbeef). In 32-bit mode. 0xdeadbeef (_end+0x9df10793): *** Invalid address 0xdeadbeef (_end+0x9df10793) -- no code -- Enter path to file 'lstat64.c':

> I don't think it's an endpoint issue. I think the SIP packet headers get > over-written by the tunnel (openvpn) protocol. I'd be rather astonished if OpenVPN itself were responsible for this. As far as I know, OpenVPN doesn't do higher-level-protocol rewriting of any sort. It just provides the "bit pipe" through the tunnel. I'd suggest several other

I'm looking for a way to delay the disconnection of a call to a SIP extension (or pad it with silence) for a few seconds, after an incoming call to that extension hangs up. Rationale: I have an Asterisk PBX (current 1.4.20 codebase), with a Leadtek BVP8051S ATA hooked to an analog phone which has a built-in answering machine. Incoming SIP connections to the appropriate extension are dialed

> In Florida some new subdivision developers have sold the > phone/cable/internet rights to a provider. They run fiber to each house > and then have the uplink to provider which isn't a traditional telco. > You can't get another provider as satellite dishes are limited in > covenants and restrictions (CCR). Those CC&Rs may very well be legally void and unenforceable.

> SIP was written in such a way that the hashes it sends for passwords > could, with only a trivial rewrite of the server code, be SHA1 instead > of MD5 -- which would increase security to the level that, currently, it > would be far more trouble than it's worth to even bother to attempt to > crack. I strongly doubt that the known weaknesses in the MD5 hash are the "weak

> In my case, I have good incoming quality and terrible quality going out. > That is, I can hear people perfectly well but they complain that my > voice drops out and is garbled regardless of who places the call. This suggests to me that you may have congestion problems in your "upstream" traffic flow. Setting QoS on the packets may not help, if whatever router you are using

>> Isn't an SSL based tunnel all TCP? > Not in the case of OpenVPN. I'm not sure about the commercial > offerings. Correct. My recollection is that OpenSSL uses TCP for the setup and management of the tunnel (e.g. authentication and key exchange) and uses UDP to carry the actual payload... each tunneled IP packet is wrapped in a UDP datagram. That way, the UDP transport

> 1 - My SIP server (Asterisk) will have some SIP clients registered in its SIP registrar. Let's say 6 SIP clients. In my project I have to implement a way of a SIP client making a call to a number and all others 5 SIP clients ring. That is, the others 5 SIP clients must receive the SIP INVITE. Can Asterisk help me with such functionality? The Dial() application lets you specify two or

> Someone on this list uses the address @sedwards.com > > I doubt this is their actual email address as there is no MX record for > sedwards.com and I can't find registration for their domain either. > > Part of my mail servers reject these emails because they cannot be > replied to, or are likely to be spam. > > Every so often I get a mail from the list

> BTW, can someone explain to a libart major like me (;-)) where echo > comes on in a telephone conversation? I seem to recall it's due to the > length of the line between the CO and the local party, but I'm not > sure. I'll try. Echo occurs when part of the signal traveling in one direction on the phone line, is reflected back in the opposite direction. It's similar

> Hmm the calls are made during the day (and sometimes very early in the > morning). Right now it looks like someone actually made these calls. If > that is the case it's somewhat comforting to know the system wasn't > compromised. However, the $25,000 phone bill still remains. Yikes. $6.25 > per minute to Cambodia seems quite steep to me. Since the Mitel had a default

> I looked at your network diagram. Try checking the configuration of the > Ethernet ports on the firewall and the Asterisk box. Make sure they are > set to auto-negotiate and not set to a fixed speed and fixed duplex. > I have found in the past that if one end of a link is expecting auto- > negotiation (as the switches probably are) and the other end is expecting > a fixed