similar to: agi debug - unable to set normal priority

Does anyone have any ideas why my recently updated * 1.2.5 system should spawn multiple * process at seemingly random intervals? Regards L:ee ########################################### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ -------------- next part -------------- An HTML attachment was scrubbed...

Hi There, I have a problem with OggEnc v1.0.2. Can you tell me, where to put or to post a bug report? The problems eare with some speciel charakters - used in Europe (Denmark, Norway, Sweden, Germany, Austria, etc..) Some Examples: Hin Vordende Sod & S? ^ 02_Bruderov paa H?gstadtun.ogg ^ 03_Huldradans - Hin Gr?nnkledde.ogg ^

Dear all, we have encounter problem when starting asterisk in the foreground, "asterisk -vvvvgc" with more 100 SIP calls concurrently. we have set ulimit to the highest value. still has this problem. Is this the problem keeping asterisk in the foreground or this is a bug in SVN 1.2 16771? Apr 5 08:48:36 WARNING[14887]: channel.c:562 ast_channel_alloc: Channel allocation

I'm using Samba 2.2.1a in a NT Network as a Domain Member. Everything works fine. But the size of Share Names from Samba is limited to 12 Charakters. Why can?t use longer Sharenames ??? Thanks for Answers

Hi All, I was able to track down what I believe is a bug when using AGI services. This bug may crash your system if your extensions.conf script is intensive in using AGI services. Depending on your system's ulimit, * keeps opening files until it reaches the system limit and then stops responding. Function app_agi/launch_script seems to leave an open and unused file. Can someone confirm this?

Hello list, what is the correct syntax ? exten => s,n,Queue(${queuename},,,,${timeout},cleanpickup.agi^${CHANNEL}) [Sep 13 10:23:58] WARNING[23551]: res_agi.c:886 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/cleanpickup.agi^SIP/329909007906-0000017a': File does not exist. Kind regards, Jonas. -------------- next part -------------- An HTML attachment was

Trying to configure voicemail with H.323 all I get is the following errors when I call 123, 666, 665, 664 or 031. I'm a newbie at this so, I think it might be a simple fix. [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper Version 0.0alpha0 by inAccess Networks

A note to all those who are avoiding writing up an AGI becasue it looks two complicated.. I have just written up my first and its awesome.. It makes Asterisk open to all sorts of possibilities.. let your imagination run wild.. I put off writing an AGI script because a) I could not find any docs b) it looked like the only way to do it was perl and I know nothing about perl and c) I am not a

Hey & a good afternoon, i installed DeusEx (u.s. version 1.0)on my SuSE. It runs good in an emulated windowmod of 1024x768. I got some renderprobs of the charakters and decided to uninstall it to check out the updated version (u.s. version patched 1.2). The wine uninstaller dont worked fine with DeusEx Code: wine uninstaller so i decided to uninstall it with the DeusEx uninstaller itself.

Hello List I'm using Samba 2.2.1a in a NT Network as a Domain Member. Everything works fine. But the size of Share Names from Samba is limited to 12 Charakters. How can I use long Sharenames ??? Thanks for Answers Joern Karthaus

hi everybody, i?ve a problem with rsync runnig. constellation: rsyncserver is cwrsync at win2k-machine (without cygwin-beta against long filename-problem), client is a linux-ubuntu box. log from the server: (i just change names with xxx, no special charakters) this error happens all the time again, /overflow: flags=0xba l1=240 l2=22

Hi, Is it possible to softhangup a channel based on priority. I mean I want to put some calls in higher priority lets say 100. If all channels are busy and somebody wants to dial an extension with priority higher than 100. How can softhangup drop a line which has priority less than 100? I will appreciate your valuable help. Thanks Smir

The Asterisk Development Team has announced the release of Asterisk 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release:

The Asterisk Development Team has announced the release of Asterisk 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release:

Hello! i'm trying to set up transfer without using the respective asterisk-function but with the built-in phone functions. my goal is to have the first callleg billed to the caller and the second callleg to the callee, who is responsible for the forward(and i can't bill a unknown caller anyways) so far it's working without problems, but my cdr's are messed. with the help of the

Hello all: This seires is an update of last version of multiqueue support to add multiqueue capability to both tap and virtio-net. Some kinds of tap backends has (macvatp in linux) or would (tap) support multiqueue. In such kind of tap backend, each file descriptor of a tap is a qeueu and ioctls were prodived to attach an exist tap file descriptor to the tun/tap device. So the patch let qemu to

Hello all: This seires is an update of last version of multiqueue support to add multiqueue capability to both tap and virtio-net. Some kinds of tap backends has (macvatp in linux) or would (tap) support multiqueue. In such kind of tap backend, each file descriptor of a tap is a qeueu and ioctls were prodived to attach an exist tap file descriptor to the tun/tap device. So the patch let qemu to

Hi, I am trying to get Asterisk 1.2.6 to run on Solaris 8 (sparc). I was able to get it to compile, but when I try to start asterisk (./asterisk -cvg from /opt/asterisk/usr/sbin), I get the following error: (snip) Asterisk Dynamic Loader Starting: [res_musiconhold.so]Mar 28 09:23:48 WARNING[18299]: loader.c:325 __load_resource: ld.so.1: ./asterisk: fatal: relocation error: file

how does a GotoIf-challenge look like to match e.g. only the first two digits? i want to strip the first two digits from an incoming pstn-call and add a zero instead so when i forward a call to a mobile the called party gets the correct number of the caller. at the moment, incoming calls from the austrian pstn are recognized as e.g. 43650123123 by asterisk, when i forward the call e.g. to a

I'm using 1.6.2 on a win 2k box. (I know I'm due for an upgrade.) I've used brute force to label a series of vectors, and I'm wondering if there's a better way to do this. I wanted to create a biplot of the 2nd & 3rd PCs, and did this: test <- edit(loadings(cv.prc.spr)) test.1 <- test[,1] test.2 <- test[,2] test.3 <- test[,3] x0 <- 0 y0 <- 0 i <-