similar to: patch for 2.4.22

Hi, Is there anyway to eliminate the color coding (for example [1;36;40m) to be stored in asterisk log file? Regards. __________________________________ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/

Hi List; I understand that I have to compile zaptel but what about asterisk? Is it enough to extract it? Well, how I will run asterisk (without compilation and installation)? Any advise? Regards Bilal ____________________________________________________________________________________ Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited

hello i'm saleh please guide me i like that i have a jmail how i have a jmail please invite me thanks bye --------------------------------- Yahoo! Music Unlimited - Access over 1 million songs. Try it free. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorbis-dev/attachments/20051016/ec04b372/attachment.htm

Anybody able to build and run 1.0.6 on Solaris? I run into "configure" problem. My Solaris box maybe a little old. I also tried to run static tinc version 1.0.4 and 1.0.3 for Solaris, but "netstat -nr" routing table won't show tun interface entry, hence ping does not work. Everything runs OK with on Linux platform though. Please let me know if anyone is running any

Dear I am very beginners for samba. I want to install samba on Debian. I want maintain file server. We have 50 PCs that are running on WinXP I want connect these PCs to one server PC it run on Debian . Over IP Range is 192.168.1.0 255.255.255.0 What are the main step to install samba on debian . DNS server is must or not? Can you help me thanks

My thumbdrive w/Syslinux and memdisk doesn't work on my Thinkpad laptop. But the thumbdrive does work on desktop platforms. And .. if i use Grub4dos with Syslinux .. it works on the Thinkpad Laptop. Any assistance would be greately appreciated! Thanks! <<john>> ____________________________________________________________________________________ Access over 1 million songs

hi, i have queries regarding openssh. 1. Openssh code is not multithread safe. scenario : - using solaris 8 for compilation & running application. - at many places fork & exec* functions are used along with other non multithread safe functions. - I have created a shared library of openssh & using it in other process - when i want to sftp

I'm getting: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 00009ms SCall: 00008 DCall: 00002 [66.98.180.77:4569] CAUSE : No authority found Under the DIDx number I'm putting IAX: default@asterisk.crystalstream.net/1567252NNNN(IAX) where N is the rest of the numbers. In iax.conf I have: [default] ; DIDx type=user

I want to add things to the prompts like: "mark urgent" "add to message" "pause while recording message" Any examples of how to do this? I'd also like to switch around prompts, not simply edit the sound files. Is it an agi, special dailplan, patching the app_voicemail.c file? All three? Any input/examples are welcome. -thanks

Hi! I am trying to setup a simple queue in Asterisk and I'm having a small problem. Our callers come in through a Bosch PBX and are immediately transferred to an Asterisk menu/IVR. If they select the option to call a SIP phone directly (eg. entering the operator's SIP extension) then the callee/operator can transfer the call to a phone within the Bosch system. What Asterisk does is

I understand this function (line 832 in app_voicemail.c) is used to retrieve a voice message. What I don't understand however is why ".txt" is appended to the end of the filename. Could someone shed some light on this for me? Thanks, Jez if (msgnum > -1) make_file(fn, sizeof(fn), dir, msgnum); else ast_copy_string(fn, dir, sizeof(fn)); snprintf(full_fn, sizeof(full_fn),

Howdy, How do you populate the queuestats from asteriskguru.com outgoing stats? Does it take an outbound dialer like Vicidialer? For good manager reports, should the user be tied to a queue and have all calls count as work? Inbound stats works well, but outgoing stats do not work and I'm not sure they should. For every Sip device, I matched the accountcode to the queuenumber (freepbx),

Hi, I recently switched to using dovecot for my IMAP server. I'm running dovecot 1.0rc10 on redhat linux 7.3. The filesystem is ext2, not NFS. I'm using mbox for mail storage. It worked okay for a day or two, but now when I try to check my mail I get an error from evolution, my imap client, saying that the server unexpectedly disconnected. In the server logs I see: Nov 2 13:10:49

Is there a way to detect Webrick at runtime? I''ve noticed a couple of differences between my dev environment (Webrick) and the production environment (Apache 2/FastCGI): * stream must be false in send_file with webrick * pwd is different - Webrick: It''s where you run ruby script/server from - Apache: It''s the public directory ... unless someone

Hello -- We have a LAN that consists of a samba domain controller and a bunch of PCs. The samba version is 3.0.20b and the PCs are running Windows XP (sp2). The Windows clients can join the domain without any problem, however, a message saying "Please wait while the domain list is created" shows up right after the first time after the clients join the domain and reboot. This

does the IPAPPEND option still exist? If so I want clearification on it please. Out of the syslinux documentation it says the following. The flag_val is an OR of the following options: 1: indicates that an option of the following format should be generated and added to the kernel command line: ip=client-ip:boot-server-ip:gw-ip:netmask ... based on the input from the DHCP/BOOTP or PXE boot

Am having a unique problem, calls received on my SPA942 seem to end after 15 seconds, but calls made from this device do not have this problem. For this device (when receiving calls) I get periodic "chan_sip.c set_destination: can't find address for host" I have set the "canreinvite=no" in the sip.conf. Does anyone have a sample entry from sip.conf for the Lynksys SPA 942

Hi guys, Here is a bit more detailed information of my problem: If I connect Asterisk PBX to the Polish telco via E1, I don't get any red alarms or anything. The line seems to be fine and the inbound calls are also accepted by the Asterisk. However, whenever I try to make an outbound call, the call is either stuck (Asterisk just displays Called g1/482.... and then nothing), or I get the

Hi all, Im network booting initrd-2.6.9-22.EL.img from pxelinux. <snip from init script> echo Creating root device mkrootdev /dev/root umount /sys echo Mounting root filesystem mount -o defaults --ro -t ext3 /dev/root /sysroot mount -t tmpfs --bind /dev /sysroot/dev echo Switching to new root switchroot /sysroot echo UMOUNTING initrd-dev umount /initrd/dev </snip> I receive error

Hi, In what Asterisk file can I edit for me to temporarily unload such modules. But later I woudl still be able to load them. Thanks Angel ____________________________________________________________________________________ Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited -------------- next part -------------- An HTML attachment was scrubbed... URL: