similar to: Sound: Recording overrun

Hello All, Finally I realised that the Asterisk demo setup didn't include support for h323. (Maybe it should have been obvious) so I went to work out how to get the h323 channel running. I had openh323 and pwlib installed as I'd been playing with vocal so it didn't take long to do cd asterisk/channels/h323; make; make install; make samples, copy the pwlib and h323 libraries to

The voicemail and voicemail2 applications both invoke sendmail as /usr/sbin/sendmail -t. However sendmail is invoked using popen(3), the input being terminated when the pipe is closed and not when a . is entered on a new line. While it looks like the code at the moment can't generate a single line with a '.', it might be prudent to apply the following patch which tells sendmail to

hello list! I am attempting to build an rpm of postfix that includes support for mysql. I've done this before with earlier versions on postfix but I am staring at this spec file until my eyes bleed and I just don't see why when I build the spec with rpmbuild mysql support isn't there. After I install the rpm I have a look at the modules as such: ldd $(which postfix) | grep -i

hello list, I'm trying to build a postfix rpm that has mysql support included. I've found the line where I need to define mysql support but it seems that I am being tripped up by some build dependencies: [root at beta SPECS]# rpmbuild -ba postfix.spec error: Failed build dependencies: MySQL-shared is needed by postfix-2.9.1-1.rhel5.x86_64 MySQL-devel is needed by

We are not having any luck with the E100p card here in Australia, it will work with a crossover cable to another device but will not talk to our Telco Telstra who probably have a weird implementation of an E1. Any suggestions on a replacement? Regards Mark McKibbin DCS Internet 64 Queen St Warragul Victoria 3820 Australia www.dcsi.net.au mark@team.dcsi.net.au Ph. 1300 665575 Fx. 1300 556595

Hello all, I require Postfix to have PostgreSQL support. Unfortunatly the RPM (nor the SRPM or it's SPEC file) that ships with CentOS doesn't include this. The SRPM (and it's associated SPEC file) that are made by Simon Mudd ( http://postfix.wl0.org/en/ ) do have PostgreSQL support, so I have downloaded that, tweaked the SPEC file and rebuilt an RPM that has PostgreSQL support.

I have asterisk configured on a machine connected to Internet by a cable modem with a public ip. The same network card has a private lan address which I'm trying to use to play with an asterisk configuration with X-lite or an softip phone. sip.conf has bindaddr=192.168.0.1 [the private address of the LAN] to which the client [192.168.0.2] is connected. Doing a tcpdump on the register

Hi, As someone new to puppet I''m trying to work out the best way to manage different filesystems and logical volumes on different servers. Specifically I would like to be able to define on a series of nodes different LVM logical volumes to create and mount. I''m trying to do this at the moment with a define of the following type: # Manage a partition and create if needed.

Two questions. First, timestamps seem to be getting screwed up when I access files on the company's NT server. Someone (using NT) creates some files. My ls -l output shows them as having screwed up dates (ie, 1935, 2027, etc). 95, 98, and NT users see the dates as normal. When I copy any files with cp, the dates change for me, although remain nonsensical, but now the timestamp is also

hi all, i tried to install postgrey for postfix and had to update postfix to version 2.1. i added the following to /etc/yum.conf [postfix-2.1-rhel3] name=Postfix Mail Server for RedHat Enterprise Linux 3.0 - $basearch baseurl=http://ftp.WL0.org/yum/postfix/2.1/rhel3/ after the update i have problems because the new version of postfix doesn?t support SASL. is there an postfix-tls rpm for CentOS 3?

has anyone managed to get these three beasties to work together ? we're using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft netmeeting default from windows xp. the symptoms are that calls from a SIP client to NetMeeting rings on NetMeeting, but upon answering the call in NetMeeting, no audio is passed between the two. eventually, the call times out and hangs up. on a

Hi all I have a problem using the Speex voice codecs when using GnomeMeeting on one side and NetMeeting on the other side. I use GnomeMeeting under Suse Linux 9.0 to communicate with a friend working under Windows XP and using NetMeeting 3.0. Under Windows XP / NetMeeting we have installed and registered the Speex voice codec. (You can find more information how we have registered the Speex codec

If you want to use the Speex ACM codec in Netmeeting, you need to register the codec with Netmeeting. Microsoft offers an API to accomplish this task. http://msdn.microsoft.com/library/default.asp?url=/library/en-us/netmeet/nm3 codec_1xk3.asp The function is called AddACMFormat in the IInstallAudioCodecs COM interface. It takes a fully initialized WAVEFORMATEX structure as well as an extra

I've been banging my head on this one for a few days and am quite stuck. I've got a gatekeeper running and everything works there. Netmeeting works calling other netmeeting clients. Netmeeting calling asterisk connects, but netmeeting can't generate the signals to make the demo do anything other than talk. But connection from ohphone always disconnects straight away. I can't seem

Hi, > I managed to get the codec into netmeeting. Unfortunately it doesn't > properly work. I tried to talk vie net, but only erranous packets are > decoded. Did I possibly register the codec wiht incorrecxt parameters or > is this a problem of the acm codec? I can't tell why. Try to register mode 8kHz Mono mode Q3 or Q4 for Netmeeting. Only these modes pack exactly 1

> Nice to hear! Do sou think you will be able to make the other modes also > compatible? But I guess these working modes are already OK for > netmeeting. I will try the Q4 16kHz mode today. Low bitrate was the design goal, not Netmeeting compatibility ;) Padding loss occurs because Speex encodes frames that do not end on byte boundaries. If you force each Speex frame to be byte

Ok I'm confusEd at this point. I have Asterisk installed with the h323 channel driver compiled and installed. I also installed gnugk. Is it needed? When I use it as a gatekeeper I'm seeing no interaction with asterisk. If I dial straight to asterisk with netmeeting I get the default extension and I can see asterisk trying to play the "congrats-demo" file but I hear

Hi, I have a little question, I use asterisk with Netmeeting client. When I call netmeeting client with a phone. I don\'t have his ID in netmeeting window i have something like : ???;..dhz instead of 28. Someone know a way to display this ID ? Thanks you so much Rattana

Hi, I'm still struggling to make netmeeting work with asterisk and oh323. I'm dialing from netmeeting into a regular phone, connected to my TDM10B. everything looks great, except that I cannot hear my voice at the FXS side, just static that increases when I speak on netmeeting's mike. Nevertheless, if I speak on the telephone I do can hear my voice on my headsets. I configured

Has anybody had this problem? I came in this morning and found my streamer down and this message flashing by on the console. "Sound: Recording Overrun" I'm running Icecast 1.3.11 and the source is Liveice on Redhat 7.1 if that makes a difference. <p>Just curious, Matt <p>--- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: