search for: ashling

Hi Tim, >> There is a distinction between logical/arithmetic shift right, but why >> not for shift left? > > The arithmetic right shift has the nice property that it preserves the > fact that x >> 1 == x/2 for negative signed numbers (unlike the > logical shift). except when x is -1. That said, this is correct: the reason for distinct signed and unsigned right

Hi, > There is a distinction between logical/arithmetic shift right, but why > not for shift left? The arithmetic right shift has the nice property that it preserves the fact that x >> 1 == x/2 for negative signed numbers (unlike the logical shift). Either because of this or because of other uses I can't think of right now almost all modern CPUs implement it in hardware, and

> From: llvmdev-bounces at cs.uiuc.edu [mailto:llvmdev-bounces at cs.uiuc.edu] > On Behalf Of Tim Northover > In contrast, it's not quite clear what an arithmetic left-shift would > be. If you want to keep x << 1 == x*2 then the logical one already > does this. Only if the left-most (shiftcount + 1) bits are all-0 or all-1. I've used machines that distinguished

There is a distinction between logical/arithmetic shift right, but why not for shift left? I'm also a bit confused by one example in the reference manual: <result> = lshr i8 -2, 1 ; yields {i8}:result = 0x7FFFFFFF Is this an error in the manual? The result is supposed to be an i8 yet a i32 is shown. -- edA-qa mort-ora-y -- -- -- -- -- -- -- -- -- -- -- -- -- -- -- -- -- -- -- -- --

Hi all, Could someone let me know the most common way that an Internet ISP would allow customers access to the PSTN?? Do they buy multiple fxo cards such as the TDM400P and rent multiple lines from a larger provider?? Would the best way be to connect to a third party voice/pstn gateway?? Is that simply a matter of forwarding all sip traffic destined for the pstn to another provider with a

Hi, First of all apologies because this isn't strictly a purely asterisk question. I am quite new to asterisk and actually to voip/telephony as a whole. I currently have sip calls working through asterisk. The asterisk server is behind a linksys router. I would now like to connect calls to the pstn. I have researched into several ways to do this but because I am not very knowledgeable about

...before. I tried plugging out the phone again but it did not make a difference. Does anyone know what those extra messages on the console mean or how I can solve this? I am obviously missing something important - How do I get it? Many Thanks. -----Original Message----- From: Aisling [mailto:ashling.odriscoll@cit.ie] Sent: 06 September 2005 18:09 To: 'asterisk-users@lists.digium.com' Subject: Asterisk BT100 Password Issue Hi, I am getting the following error when I attempt to listen to voice messages by dialing 9999 (I can hear nothing): --Executing VoiceMailMain ("SIP/2092...

...ow mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. ---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW: SER Asterisk Voicemail Date: Thu, 10 Feb 2005 16:45:53 -0000 Hi all, I have SER and Asterisk set up together with ser handling user registrations and asterisk providing voicemail services. When I ring a phone and it doesnt answer a...

.../var/spool/asterisk/voicemail. The default context resides there but new ones are not created. Has anyone ever experienced this or does anyone have any idea as to how I would solve this? Hope someone can shed light on this, Many thanks, Aisling. -----Original Message----- From: Aisling [mailto:ashling.odriscoll@cit.ie] Sent: 07 September 2005 13:54 To: 'asterisk-users@lists.digium.com' Subject: Eeven Stranger - Asterisk BT100 Password Issue Following on from my below email, things have taken another bizarre twist.. I have continued getting the error when 2092 tries to listen to mess...

Hi guys im trying to get NAT working on my system. im using 3 phones, 2000 = xlite, 2001 = xlite, and 2010 = some piece of crap voip phone. when i ring from anywhere to anywhere u can either never hear anything on both ends, or just 1 end can hear stuff. below is the output of sip show peers 2010/2010 203.1.68.90 (D) 255.255.255.255 49534 UNREACHABLE 2001/2001 203.1.68.90

...idn't allow me to interrupt the menu. [incomingpstn] exten => s,1,Wait(1) exten => s,2,Background(MainMenu) ;exten => s,3,WaitExten(10) exten => 1,1,Goto(internalExt,s,1) exten => 2,1,Goto(mainconfmenu,s,1) Many Thanks, Aisling. -----Original Message----- From: Aisling [mailto:ashling.odriscoll@cit.ie] Sent: 11 January 2006 10:14 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main Menu Hi Kokmeng, Unfortunately that's wasn't it. WaitExten was executed but then I still get th...

...to:asterisk-users-bounces@lists.digium.com] On Behalf Of Giovanni Miano Sent: 05 January 2006 21:09 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Incoming PSTN Calls Is Exist "InternalExtension" context ? and 2093 exten ? 2006/1/5, Aisling < ashling.odriscoll@cit.ie>: Hi all, I am having difficulty getting incoming PSTN calls working. I have set up an account with a third party provider. In my system, the user register with SER and use Asterisk for PSTN access, voicemail etc My provider told me to change my sip.conf as follows registe...

...rade". However I get an error: Makefile:16: *** missing separator. Stop. Make[2]L Leaving directory '/usr/src/asterisk' Make: *** [depend] Error 1 Has anyone come across this or does anyone know a way of solving this? Many thanks -----Original Message----- From: Aisling [mailto:ashling.odriscoll@cit.ie] Sent: 26 August 2005 15:44 To: 'asterisk-users@lists.digium.com' Subject: cvs update error? Hi, I'm experiencing a problem with playing back my voicemail. (Failed to write frame). It has been indicated in the archives that this is problem can be solved by updatin...

I wonder does anyone have any thoughts or can give me some direction on the following: I have an asterisk testbed environment set up. My task is to make a personal number service available whereby users would be given one number (perhaps a voip number) and this number would enable them to be reached via the pstn, pots, gsm etc.... Does anyone have ideas where I could start looking at sites to

Hi, Hope someone can help me. I am a student who hass been given a project to develop a dynamic bandwidth service. I currently have a linux router which at the moment gives users static bandwidth and assigns each of them to a bucket. I have not gotten information yet as to whether each bucket is serviced in a round robin fashion or whether certain buckets get preferential treatment (i.e. bucket

Hi all, I am having difficulty getting incoming PSTN calls working. I have set up an account with a third party provider. In my system, the user register with SER and use Asterisk for PSTN access, voicemail etc My provider told me to change my sip.conf as follows register => username:password@sip.blueface.ie/2093 ; To receive incoming calls specify this block and

Hi, I am getting the following error when I attempt to listen to voice messages by dialing 9999 (I can hear nothing): --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. I read in previous posts that this may be to do with the dtmf

Hi, I'm wondering if anyone has experienced an issue with the XLite softphone and asterisk accepting dtmf? I can listen to my voicemail perfectly from my hardphone. However when I dial the voicemail number from my XLite softphone and enter the password at the voicemail prompt, an error appears vm-incorrect and I get an "Unable to read password" message on the asterisk console. Has

..." HTH ------- Ing. Julio Alvarez Tejera Unix Trends *BSD, Solaris & Linux VoIP & CT Solutions Finder Asterisk PBX Consultant Costa Rica Land +506-359-9753 USA Toll Free +1-888-899-6269 --------------- "extremely stable systems" ----- Original Message ----- From: "Ashling O'Driscoll" <ashling.odriscoll@cit.ie> To: <asterisk-users@lists.digium.com> Sent: Thursday, January 13, 2005 10:57 AM Subject: RE: [Asterisk-Users] SER vs Asterisk for SIP >From my (fairly limited) understanding, I think the fundamental difference is that Asterisk is a p...

Hello, Hope somebody can help me - Asterisk is behaving very oddly and I'm totally stumped! I have SER and Asterisk running on the same box. I want SER to listen on port 5060 (it is) and Asterisk to listen on port 5062. I have configured my phones to register with x.x.x.x:5060 (SER) and Asterisk will purely act as a voicemail server at the moment. However I cannot get Asterisk to listen on a