similar to: Asterisk as a distributed paging system

Hi, I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone

Hi There, i think all VOIP clients for Linux are unusable! i got testet: Linphone + Linphonec all in version 12.2 Kphone gophone and other... the only programm that is usable is gnomemeeting... does anybody knew some other tools? Best Regards, Mark

Anyone have some recommended equipment for alerting people to calls in a noisy environment? I have Polycom IP550 phones set up in some really noisy environments - our mine hoists - and they tend to drown out the ringers. I'm using Clarity WR100s now. They're analog devices, attached to Linksys PAP2T ATAs as part of a call group to get a loud (advertised as 95dB) ring out there, but it

Hi, I'd like to implement some way of controlling remote SIP clients while in a call, to execute remote commands. The call topology (think of a PA system) is this: * the caller is in a MeetMe() conference room * the callees are Page()d, then the dynamic conference room is connected to the previous one I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the caller to the

I am dying here with linphone (not sure if it is crap software or just me being an idiot) but out of the box debian installations of two linphones fail with a "Got SIP response 415 "Unsupported Media Type" back from 192.168.1.3" Can anybody recommend a particular SIP soft phone that broadly satisfies the following criteria? 1. Run on linux. 2. Simple to use and setup. 3. Is

I'd like to cobble together a videophone from an analog phone, connected to an Asterisk FXS channel, and a co-located video camera, connected to a video grabber card on the Asterisk server (so I have a Linux video device providing the video stream). When a call is made from the phone, I'd like to somehow add the video and produce a SIP video call. I don't want to use any sort of

hi all, please help... I am calling in the simplest way among two linphone clients connected to one asterisk server... the call ends on one side without any sign of problem, while on the other side it stays connected. I checked the SIP dialogue and at some point the server sends a BYE message to one party I have no timeout set, though the duration of a call is always around 20s. the two

I'm trying to improve Opus on an AMD Geode CPU, which has limited SSE support (called 3DNow!), but MMX. Without optimizations I can only encode 16 bit audio @16KHz with complexity up to 2-3 without underruns. I tried compiling with SSE2/4 optimizations, but all I got was a crash with SIGILL, so I looked into optimized code and found that a good starting point was the dot product, so I

Hi Guus, hi all, please find attached a proposed feature implementation for tinc. As mentioned in http://www.tinc-vpn.org/pipermail/tinc/2010-May/002344.html , my goal was to connect nodes on the same LAN using their local (LAN) endpoints. I've implemented a multicast sender, which announces its own endpoint on every connected interface regularly. All nodes receiving multicast

We currently have an Asterisk system that is hooked up to our old paging speakers via sound card, plugged into two amps. Each amp drives up to 8 analog speakers in each warehouse (we have 2). Both warehouses are around 30k square feet. Both have a large number of printing presses. The computer system is that is running Asterisk is around 10 years old and starting to fail. I'm looking to

HI Guys, I am trying to use the RTPPage application on asterisk 1.4 using the Snom 320's?? My goal is to do the paging using a multicast IP address. I tried the app_rtppage.c and i can only do unicast on the snom's and i was unable to do a multicast. https://issues.asterisk.org/view.php?id=11797 http://svnview.digium.com/svn/asterisk?revision=101218&view=revision My dialplan

I know this may be a bit off topic... I'm trying to play a pre-recorded message to a group of Aastra phones using multicast paging. I can page phone to phone without issue, but sending from one of my servers to the phones results in garbled audio. Anyone else been able to make this work without problem? My VLC command line is below. cvlc -v emergency-test2.wav --norm-max-level=5 --sout

Hi, I ran "yum update" this morning on a 7.2 machine with all of the cr updates applied to it and yum wants to install lvm-cluster and a bunch of deps. Dependencies Resolved ====================================================================== Package Arch Version Repository Size ======================================================================

Hi All, Is it a way to prevent joining a domain with a netbios name that already used by other domain member?. For example, if I have SAMBA server "SA1" already joined a domain and then I go to a different SAMBA server and make it join the same domain with the name "SA1" also. What I found out is that the domain controller does not care and it will let me join the

Hello, Anybody now how make with puppet dependence on other service on another node? like : require => Service[dbnode:postgresql] I need ensure that the db is running on another node before start my appserver. Thank you. -- You received this message because you are subscribed to the Google Groups "Puppet Users" group. To post to this group, send email to

Hi, How can I have vi with syntax hilghting for root ? Regular users have it, but not root's. I've seen the hidden files of a regular user home, but found nothing. Thanks in advance. Warm Regards, M?rio Gamito

Hello listmates, This is not specifically CentOS-related - though I will probably execute this design on CentOS if I decide to do so. It will certainly be some kind of Linux. At any rate, here's my situation. I would like to build a fairly large storage solution (let us say, 100 TB). I want this solution to be distributed and redundant. I want to be able to lose part of the machines involved

Greetings everyone, I was wondering if there was a way to change the codec that Asterisk uses when streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream. In the CLI, when I have a multicast stream in progress, I am typing 'core show channel MulticastRTP/0x7f7........' to get lots of helpful information. I have noticed that when I do a MULTICAST page and send data

Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem

Hi Yves lavaan looks like a very nice package. From the tutorial introduction I see you create path diagrams for some of the models you describe. How did you do this? I don't see a function for this in the package. I know there is a path.diagram function in the sem package that uses dot to draw the diagram, but I've always found the layouts from dot somewhat strange for path diagrams