search for: screeches

Hi all, We have just set up trixbox latest with a Rhino r1t1 card, hooked up to a plain E1 PRI line. We call fine SIP to SIP, but as soon as we make a call from SIP to PSTN all sounds become unintelligible screeching or static kind of noise on both ends, when we call PSTN to SIP the PSTN side seemingly OK at least we hear no screeching sound, but the SIP side is a even worse screeching

Asterisk SVN-trunk-r7353M (will be moving to 1.2.2 this weekend) E1 connected to Sangoma A102 SIP phones (Cisco 7960) I've been making a call from my mobile to the office, when, suddenly the conversation is terminated and replaced by a "fax-type" sound. This has happened to me several times over the past year, so it's not the version of asterisk (we've had cvs trunk and

Hi, Is there any native software for OS X to convert an audio file to speex? Even a command line would be great. I looked for hours and couldn't find a program. I tried VLC but it just outputs screeching noises. Sincerely, Patrick Krol patrick9@net-link.net

I'm sure dumb when it comes to describing things that happen on my system. I'm making an outbound call on my ATA186 when another call comes in. I first get the nasty CID screech and then the periodic call-waiting tone. So far, so good. Then I hookflash, and just like it's supposed to, the waiting caller is on the line. But during the duration of that conversation, my console

In the simplest terms I can think of, I'm going to describe what I want to do and I want to know if it's possible in the current version of asterisk. Can I take a T38 call from an ATA, convert that back to analog and have asterisk screech that out on a POTS line to a remote fax machine. Would it work? And could I receive an incoming fax the same way? Please don't talk to me

Bear with me this is a bit long winded. I am having some issues making automated outbound calls over Broadvoice from my Asterisk 1.4.2 server. For reference, none of the below issues happen when I make the calls to VoIP phones attached to the Asterisk server. What I am trying to do is call, using a .call file, out via the SIP trunk we have setup, and when the party picks up use AMD to

I just deployed * on my home system last Sunday. 2x since then the Zap hardware seems to have malfunctioned on some way. One time it would just screech out one FXS, even though it would ring. The other time * would bridge to my FXO but it never got out on the line. I have a new TDM400 with 3 FXS and 1 FXO. Both times I tried unloading the zaptel drivers (which worked) and reloading them,

Hello all, First, the vital stats: FreeBSD 6.1-STABLE, dovecot 1.0.rc7 from ports. I have run into a vexing little problem that has brought my deployment of a shiny new dovecot server to a screeching halt. Using 'periodic' I've configured regular backups to occur using 'dump'. I have a monthly level 0, a weekly level 4, and a daily level 8. These dumps take advantage of

Hi, I've googled it and look in voip-info.org <http://voip-info.org> without any success. Hope someone can point me to the right direction. I saw a couple similar questions, but don't see any answers. Fedora Core 2 2 X100P(clone) PSTN Asterisk 1.07 Everything seems to be running fine, but on occasion, Asterisk answers the call with high pitch screech sound (like fax or modem

When I make a regular call from my SIP phone connected to my Asterisk server I have no issues, however when I make a call using Originate : 'Channel'=>"SIP/1XXXXXXXXXX@sip.broadvoice.com", 'Context'=>'mycontext', 'Exten'=>'899', 'Priority'=>1, 'Callerid'=>'whatever')); It creates a screech sound when the

Hi All, I know that this may not strictly be an asterisk issue, but if anyone has any ideas on this prob it would be appreciated. I have a working asterisk setup using Suse 9.1, a BT ISDN BRI card (Fritz?), sipura 3000 and a few cisco 7940's. I wanted to do add faxing into the equation, so I thought I would try and setup the following: PSTN->asterisk->hylafax->pc At the moment

Not sure if this document will be shown to any dtrace users but here is a link if you want to comment on it http://sources.redhat.com/systemtap/wiki/SystemtapDtraceComparison I have went line by line and commented on its content and in accuracies and posted it both to the systemtap email list and my blog, you can see the blogged version at

On Thu, Oct 22, 2009 at 3:54 PM, Dan Gohman <gohman at apple.com> wrote: >> Anyway, after thinking about it, it should always be safe to override >> the Module to remove the target data pass, even if it isn't safe to >> override the Module to substitute different target data.  But I still >> think you should at least have the option to supply target data >>

We recently tried the upgrade to 2.0 Beta series (just finished testing Beta 4) on our network. Everything works fine with 1.9.18pl10, but once we start 2.0 in its place we start getting the following in the log files: [1998/12/17 05:00:12, 1] smbd/files.c:file_init(219) file_init: Information only: requested 10000 open files, 246 are available. [1998/12/17 05:07:52, 1]

On Mar 16, 2005, at 11:41 PM, Jean-Marc Valin wrote: >> Is there any native software for OS X to convert an audio file to >> speex? Even a command line would be great. I looked for hours and >> couldn't find a program. I tried VLC but it just outputs screeching >> noises. > > The standard Speex distribution includes the speexenc utility, isn't > that what

Hmm.. just switched back to icecast hoping that 1.3.9 would have fixed some of the bugs I was running into, such as icecast gobbling up more and more memory until it would tank around 200M or so (in a box with 256) if left running a few days, and got some general questions for anyone in-the-know. ;) 1. Is there a definitive list of what players on what platforms support the x-audiocast

I'll admit I am new to LVM2, but I have got myself in a bad spot. I renamed the LVM volume and volume group so that I can keep track of what is in them. I have changed grub's menu.lst, /etc/fstab, and /etc/mtab, but somewhere else there is still something telling lvm that my root drive is on VolGroup00. Where is it, and how do I convince it that VolGroup00/Volume00 (or whatever the

I received an additional module for my TDM400P yesterday. I installed it last night. Before I installed it, I had a working 1x1 PBX. After installing the physical hardware, I spent a couple of hours unsuccessfully attempting to get dial-tone on the new FXS part. I gave up, made sure the old FXS still rang and went to bed. I should have tested further. This afternoon, I got a call at work

I've been trying to get a console channel working without success. The sound card, which is built into the motherboard, is a VIA Technologies, Inc. VT82C686 AC97 Audio Controller. Using the oss drivers (vi82cxxx_audio) in kernel 2.4.23 and chan_oss, I just get beeps and screeches. Using alsa drivers (snd-via82cxx) and chan_oss (using the alsa oss emulation), playing sound works, but anything recorded via the mic has a weird scratchy echo over it. If I use a program other than asterisk to record (such as rec or arecord), the recording sounds fine. It works even if I'...

Has anyone used this feature successfully? I 'think' I have a .wav file that it wants. Here is what 'file' says: sf-george.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz I see the logs on my web server as it tries to access it, but all I get is a screech out of the phone.