similar to: This needs to be easier

how do i troubleshoot no ring tone. It was working and all i added was the lines below now it doesn't ring. Edit sip_nat.conf for proper NAT: localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here) externrefresh=10 fromdomain=DOMAIN.com (Set your external domain name here) nat=yes qualify=yes canreinvite=no Add extra codecs to

I understand that the action of pxelinux has changed so that when "quiet" is specified in append you no longer see the progress of the kernel and init image download. My question is, is it possible to have pxelinux be verbose (ie: show progress), but then pass quiet to the linux kernel so it is quiet. I run a downstream project that uses pxelinux and I wanted to know if this is

Safe Harbor Games (http://www.safeharborgames.net/) is a popular site in the backgammon community. I am one of several people trying to get the game server to work on Linux with Wine. I am using a Asus EEE 1000HE with the latest version of Easy Peasy (Ubuntu) and Wine. The game software requires browser to launch and I was successful in getting this to work by loading windows version of

i have posted this before but was unable to resolve it. i have some new info so i figured i would try again. the trace from bandwidth.com are below. they are telling me that the ip that is bold should be our ip not bandwidth.com. i have changed every setting that i can see and nothing fixes this. Where would i change this at? they cannot tell me. INVITE sip:+185993133333 at 216.82.224.202

How can i activate "ChanSpy" to spy on a dedicated extension? I see the following in "/etc/asterisk/extensions_additional.conf" [chanspy] include => chanspy-custom exten => 501**,1,Chanspy(801) exten => 501**,n,Hangup exten => 502**,1,Chanspy(802) exten => 502**,n,Hangup But when i try to call "501**", it doesn't give any response. Thanks.

my problem is this. I have google forward the call to gizmo5. I have this line in my sip file : register => user:password at proxy01.sipphone.com I believe this lines connects asterisk with gizmo5 so when it gets a call from Google, asterisk will answer it? At the end of my sip file i have this [Calls-From-Gizmo-Network] type=user context=demo disallow=all allow=ulaw allow=ilbc allow=gsm

Hi, LLVM documentation is not clear. Is it possible to write a machine pass? I am trying to insert some machine code before the return instruction. ideally, I'd like a pass that runs the last one before generating assembly. How can this be done? Thank you, Dan _________________________________________________________________ Hotmail has tools for the New Busy. Search, chat and

Suddenly i found an error while booting, it says: Fuck: can't open /dev/kmem for read/write (2) So this is why, the Asterisk and Zaptel can not start. Any Suggestions Please Thanks a lot Torintino _________________________________________________________________ Windows Live: Make it easier for your friends to see what you?re up to on Facebook.

I found "7777" and "*65" are not working. Please how can i re-enable them again. Thanks _________________________________________________________________ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you.

Hi, I am looking for a hosted / virtual IPBX *PLATFORM* for service provider.Such hosted IPBX platform is aimed to be as a service, so that final customers don't have to install, maintain, or upgrade any PBX hardware or software. It should have a control panel for end users to create / edit extension, conference rooms , IVR menus, etc Could you please indicate companies that offer such

Greetings List. I'm facing a strange issue with one of my providers.. after sending an INVITE request my server places the call on hold.. until the call is answered.. this is happening only with this provide although i have 3 other providers i route calls through.. can anyone explain what is going on? -- Tarek Sawah / Integrated Digital Systems / CCNA, MCSE, RHCE, VoIP / +1 347 562 2308

Greetings List,Trying to interconnect with a new provider.. the require a?compliance?with?RFC 3261 ?so knowing less than needed about RFC documentations.. i would like to know if Asterisk is actually in compliance with?RFC 3261 or not..?Can any one help with this? Regards -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308

i own a CyberPower OR2200LCDRM2U and was wondering if any nut drivers support this ups. none i have tried have worked although usbhid-ups did tell me that the productid=0601 was not supported yet. i see in the compatability list on the nut website that the PR2200 is supported with the powerpanel driver but the OR2200 is not listed. i am interested in what i am doing wrong or if this is a support

John M. Chambers Statistical Software Award - 2010 Statistical Computing Section American Statistical Association The Statistical Computing Section of the American Statistical Association announces the competition for the John M. Chambers Statistical Software Award. In 1998 the Association for Computing Machinery presented its Software System Award to John Chambers for the design and development

I have setup my asterisk box using freepbx. I can call extension and make outbound calls. the outbound calls drop between 10-30sec. we are using bandwidth.com and they have logged our call. below is your bad followed by what they say is a good call. I can't figure out where the problem is on your end. I know we are missing some stuff at the bottom but I don't know where to start.

Hi ALL... I want to know if the files in /Maildir/cur & /Maildir/.Sent are modified by dovecot. I have a script which calculates the age of the mail files. I think that if they are continuously modified by dovecot I won't be able to calculate the age of the files. Can Anyone plz advise on this.... Thanks CoolAtt

Sorry if that's a vague subject, but this problem is a little weird and I'm just wondering if there are any suggestions out there. We've got a Samba server (3.0.23) running on a CentOS 5.3 server offering up a data share of 7TB on an XFS filesystem. The authentication all happens through a Samba PDC with an LDAP backend all on a different server. The system in question is just a

I recently did a set up where I replaced a simple D-link home router that was having trouble processing a T1's worth of bandwidth with a linux machine running iptables. the kernel was 2.6.29-r5 and I chose the SIP connection tracking modules from the menuconfig. Router worked fine for normal traffic, but I was unable to get the SIP phones to work. Using ngrep it was plain to see

Not sure if this is a bug yet, so I wanted to ask around to see if anyone else was having this issue. I have pollmailboxes=yes set in voicemail.conf but externnotify is not called. I know it isn't the externnotify script because if the changes are done in asterisk then it is called properly, if the changes are done via our webserver then it is not. Also, we use odbc voicemail storage. Thanks

Hi, I need to perform some transformation after code was lowered. I think I need to do this in llc and use a MachinePass. However, it is unclear how. I am able to change the register allocator: llc -f -load XXX.so -regalloc=xxx foo.bc but I don't know how to just insert a different pass and pass new command options to llc. I found plenty of documentation about 'opt', but little