similar to: Realtime problem host='dynamic' in 1.2.26.1

Hello. I would like to run asterisk as an user other than root. I have seen some tutorials on the web, but I would like to know if there is some ?official? how-to for this. Is there? I looked at a thread on reviewboard regarding this (https://reviewboard.asterisk.org/r/654/). It was Paul Belangers work trying to make the installation process take care of this. But the conclusion seem to

Hello, We are setting up an asterisk system for voicemail with video possibilities. We are not using the voicemail app, but rather writing our own dialplan logic. The part of recording, and replaying, the voicemail works, and we receive both an h264 and an wav-file. What I now wonder is how to convert these into one file playable by a (standard) media player. I have not found any real good

Hello, I have written an AGI script for asterisk that uses google translate for text to speech synthesis. It supports a variety of different languages, local caching for the voice data and wideband audio. The voice in most languages is female and the quality of the synthesized speech is very high. More info about the script can be found here: http://zaf.github.com/asterisk-googletts/ the first

Hello I have a problem where I would like to be able to send an arbitrary SIP domain when sending a call to a registered friend. By default the from domain is set to the IP of the Asterisk server, but I would like to set it to something else. The case is that when a call from a foreign domain comes in to the Asterisk, it will connect it to the callee (but with the domain changed). When

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

Hello! I am going to use the JITTERBUFFER function in a SIP (and local channels) only setup, but have some questions of how to use it: 1. Do I need to activate jbenable in sip.conf? Or is it enough to call the JITTERBUFFER function? 2. What is the preferred way to invoke this function? Say I have channel A which is not in need of buffering, while channel B do need it. If A

Errors keep coming back when I try to configure WINE. This is on RHEL5, and I do not have sound support installed (if it makes any difference). [root at dev02 ~]# winecfg wine: creating configuration directory '/root/.wine'... ALSA lib seq_hw.c:457:(snd_seq_hw_open) open /dev/snd/seq failed: No such file or directory Could not load Mozilla. HTML rendering will be disabled. X Error of

Is there anything special needed to load up a TDM10B card?? I got the card today.. Took it from the box, put it into a PCI slot.. connected the power to the card and booted the PC.. I have removed the X100P to avoid confusion and I have the following in the config files.. in /etc/zaptel.conf # For the X100P #fxsks=1 # For the TDM10B fxoks=1 in /etc/asterisk/zapata.conf [channels]

> > 1. Do I need to activate jbenable in sip.conf? Or is it enough to call > > the JITTERBUFFER function? > > You only need to use the JITTERBUFFER function. > > The jbenable option will enable a jitter buffer on every channel > created for that peer (or, if global, for every peer in the system). > Depending on the version of Asterisk, it will also place the

Hi all, I am trying to send an extra header in SIP INVITE Message , i.e (email="me at me.com") but when I check the Message at the target that header is not there So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message? Thank u

Hey, all. I'm currently trying out GlusterFS 3.3. I've got two servers and four clients, all on separate boxes. I've got a Distributed-Replicated volume with 4 bricks, two from each server, and I'm using the FUSE client. I was trying out failover, currently testing for reads. I was reading a big file, using iftop to see which server was actually being read from. I put up an

I've made packages for Debian Squeeze for Gluster 3.4.0-alpha2, they are available on http://torbjorn-dev.trollweb.net/gluster-3.4.0alpha2-debs/. They built and installed successfully, and have been running nicely for a couple of hours, but your mileage may vary. The Debian packaging is on http://torbjorn-dev.trollweb.net/gluster-3.4.0alpha2-debs/glusterfs-3.4.0-debian.tar.gz. I took the

Dear list, I'm looking for a function to generate (simulate) a random Weibull point process. Can anyone help? Cheers, Torbj?rn Ergon, University of Oslo

WTF is a jitterbuffer? Sent from my Verizon Wireless 4G LTE smartphone -------- Original message -------- From: Matthew Jordan <mjordan at digium.com> Date: 01/29/2015 10:41 AM (GMT-05:00) To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] JITTERBUFFER function On Thu, Jan 29, 2015 at 4:56 AM,

[1.3.20]-Capture The Mouse Works Great. Suggestions... Hi, I've been crying for a long time about mouse issues while running a full screen game in an emulated desktop window, but with newest Wine these issues are gone. Would it be possible to have some hot key that the user can press to uncapture the mouse? For example, VirtualBox toggles mouse capture by pressing the right [Ctrl] key on

I'm using a cisco 1760 with a VIC2-4FXO card for my calls to PSTN. If a user on a softphone hangs up first the PSTN port on the cisco is released and new calls can be made on the same voice port. But when the user on the PSTN side hangs up first the voice port on the cisco stays open until the user on the softphone hangs up. Any ideas what I'm doing wrong?

Hi guys I have met some troubles when I want to evaluate the glusterfs performance with small file workload. 1: What kind of benchmark should I use to test the small file operation ? As we all know, we can use iozone tools to test the large file operation, while for the sake of memory cache, if we testing small file operation with iozone, the result will not correct.

Hi, Is it possible to limit the maximum number of IMAP connections allowed for a user from each IP address, on the proxy server instead of on the mail store server? mail_max_userip_connections works well when the client is connection to the mail store without proxy, but when using proxies the POP/IMAP server will register the remote IP (rip) as the proxy server's IP address - thus a low

Hi, I am running Redhat, I loaded the mysql and mysql-devel RPM's and then recompiled *.. I thought it would be that simple but it looks like I have missed something becasue it doesn't look like the module has been complied.. What did I leave out? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by

Hi. With SCP, it seems like the option precedence is ignored. Although this seems to work well with SSH. $ scp -oStrictHostKeyChecking=no hs21-dev04:/tmp/1 hs21-dev02:/tmp/2 The authenticity of host 'hs21-dev04 (192.168.12.11)' can't be established. RSA key fingerprint is ec:0f:eb:b2:fa:6f:50:ef:89:64:01:5e:c9:cc:54:20. Are you sure you want to continue connecting (yes/no)? $