similar to: Installation on CYGWIN Failed (PR#9442)

Dear Xen community, I have Xen 3.0 installed on RedHat Linux Enterprise RHEL4U2. "xend install" runs fine with no error messages. However, when I start "xm cr guest-vmx.conf" I do not get any new window open for the new guest OS. "xm list" shows that the vmx has started and seems to be working fine (just for testing, when I type "xterm" an X window

Hello, I have a problem with connecting a Digium X100P card to a Brazilian analog line. Can somebody help me out with this problem? My /etc/zaptel.conf is loadzone=br defaultzone=br fxsks=1 My /etc/asterisk/indications.conf [general] country=br [br] description = Brazil ringcadance = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 congestion =

Folks, I'm fairly new to this whole Linux UPS thingie, but I'd quite like to have a look at getting my UPS to work under Linux and would be grateful for any help in getting a driver. I have a reasonable working knowledge of Linux and software development, and thus am happy to modify config files, alter kernel settings, etc, although I'm no C guru. I have a Kebo UPS-650D, which

I'm filtering that is being deliverd to postfix mail server with amavisd-new . I want spam with spam f level 1 - 8 to ad a tag any everything above to be delete is this posebol? If yes how? Met vriendelijk groet, Bas van Dikkenberg GISkit bv BFVD1-RIPE Tel: +3130-6340430 Fax: +3130-6342433 Prive Tel: +3130-6372769 Mob:

I have a question. I have two numbers on Asterisk like 902121234567 and 902123645789 and i want to divert first number's call to Trunk if second number is unregistered. Is it possible? ?f yes, how? Flow Diagram: *Two numbers are registered on Asterisk 902121234567---------------------------- registered to Asterisk

Hello, I am new to Asterisk and new to this list. I got Asterisk setup and running using Asterisk@home, and purchased a PolyCom SoundPoint IP500 phone to test out. I cannot get the phone to talk to the Asterisk box. On bootup of the phone, it tells me that it cannot contact boot server. Why is that? It gets an IP fine, and I have also tried manually setting the IP of the phone and the Asterisk

Hi all, I need help. I have a Linux box with SER as a proxy server with ip phones attached on it , and another linux box with Asterisk and T410 card connect to an E1 line .Whenever there is a call from PSTN it is passed to Asterisk and then to SER box and then to the phone .every time an invalid number dialed from PSTN to SIP phones connected to SER asterisk says that the call is progressing

Hi All: I'm a newbee to Asterisk. I currently working on a project and want to know if Asterisk does support R2 Signaling. Thanks Begra8fl >From: asterisk-users-request@lists.digium.com >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs >Date: Tue, 04 May 2004 13:32:00 -0500 > >Send

Hello, I'm looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/

Is there an up and running provider of SIP termination in Brazil? I know that there are some people building on a SIP termination solution. But who as it up and running ? Best regards, Han -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040921/f1043e19/attachment.htm

Hello - My single line extension users (connected via channel banks) need to be able to hang up faster. If they just flash the hook it doesn't disconnect right away. Any ideas on how to resolve this? Thanks, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am using asterisk 1.4.17 which is connected to a SIP trunk supporting rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for SIP clients I have set dtmfmode=info. So when I make a call to a cell number using the sip trunk and then press digits I can see the 2833 dtmf events coming to asterisk

Hi, I tried to compile R-2.10.0 src on 64 bit window. After install Rtools and wingw-w64 compiler and put it the first of my PATH variable, but i got the following error when i tried to compile. Anything I missed? thank you. x86_64-w64-mingw32-gcc -std=gnu99 -I../include -I. -I../extra -DHAVE_CONFIG_H -D R_DLL_BUILD -O3 -Wall -pedantic -DR_ARCH='"x64"' -c malloc.c -o

So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither

Happy holidays! I am experiencing an issue when trying to check my mail using IMAP. with Dovecot I have tried checking my mail using a full GUI client (Thunderbird) and telnet. Both times I get disconnected before all of my messages can be downloaded and I see an error in my mail log. Here are the details: [root at cust19-1-prod-domain userqa]# dovecot --version 2.0.9 [root at

Bruce et al. I'm posting a new thread with the "Top Posting" subject so I won't draw complaints about "hijacking" the 4-port thread. Top Posting refers to the practice of sending a message with a reply at the top and including the entire thread below the reply. I prefer this. If I'm actively following a thread, the most-recent information appears at the top

Has anyone seen good scripts or documentation on Voicemail groups? We are looking to have a system where you can send a voicemail to multiple mailboxes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060518/777a7b83/attachment.htm

asterisk linkedin group I have created an asterisk linkedin group for anyone interested. http://www.linkedin.com/e/gis/45252/66270A773F53 Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Board member of Connectech Greater Detroit www.connectech.org ________________________________ Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph.

Does anyone have any experience using a PRI gateway, I am looking for a way to have multiple asterisk boxes use one PRI, and send that over the network. I herd there are copper gateway devices (like a X100P card, only it registers with asterisk using sip, and it doesn't have to be physically connected to the box) Does anyone have any experience with a PRI gateway? And could tell me the cost

Hi all. I posted this earlier but never got any advice that helped. If anyone knows how to get this going, I'd appreciate some advice. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten =>