search for: x205

http://grandstream.com/BETATEST/HT488_496_386/

Hi all I am in the process of upgrading my XEN servers from 2.4 to 2.6 kernels as I am having better results on 2.6. However one machine (the big daddy) refuses to boot a 2.6.9 xenolinux kernel. It''s an IBM eSeries xServer x205. I have selected what i think are the appropriate options in menuconfig but it refuses to boot into dom0. By refuse I mean it initialises the hypervisor, gets to loading the dom0 kernel and immediately reboots. There''s no error messages and it doesnt even get to Scrubbing memory. This ma...

...dat1$X157 <- as.factor(dat1$X157) > dat1$X158 <- as.factor(dat1$X158) > dat1$X162 <- as.factor(dat1$X162) > dat1$X169 <- as.factor(dat1$X169) > dat1$X200 <- as.factor(dat1$X200) > dat1$X202 <- as.factor(dat1$X202) > dat1$X203 <- as.factor(dat1$X203) > dat1$X205 <- as.factor(dat1$X205) > dat1$X206 <- as.factor(dat1$X206) > dat1$X209 <- as.factor(dat1$X209) > dat1$X210 <- as.factor(dat1$X210) > dat1$X225 <- as.factor(dat1$X225) > dat1$X269 <- as.factor(dat1$X269) > dat1$X283 <- as.factor(dat1$X283) > dat1$X290 <-...

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems

...lock, label %if-false-block if-true-block: ret i32 0 if-false-block: ; Commenting the following two lines will avoid the bug %x198 = getelementptr [13 x i32]* %texture.row_stride, i32 0, i32 0 store i32 1, i32* %x198 %x204 = getelementptr [13 x i32]* %texture.data_ptr, i32 0, i32 0 %x205 = load i32* %x204 ret i32 %x205 ; CHECK: # %if-false-block ; CHECK-NEXT: movl $1, 16([[REGISTER:%[a-z]+]]) ; CHECK-NEXT: movl 120([[REGISTER]]), %eax ; CHECK-NEXT: ret } ;; Optional main function, to used with bugpoint ; ;declare i32 @printf(i8*, ...) ; ;@fmt_str = internal constant [10...

...server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote server. Thanks, -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856

I am getting a TON of these errors on the console. I Googled and wikied and greped found the error in the source but cannot understand why it is happening. The system works fine, no dropped calls, no echo, it will even run for weeks with this error. But it just scrolls and scrolls on the console. Temporary fix was to turn off the console monitor! :-) Any ideas. Apr 16 10:40:12

...ma.table". So it appears there is something wrong in some place dealing with understanding the columns for tables in schemas. Any ideas? Any help would be much appreciated. Thank you. Ben Stabler Project Manager PTV America, Inc. 1128 NE 2nd St, Suite 204 Corvallis, OR 97330 541-754-6836 x205 541-754-6837 fax www.ptvamerica.com Ben Stabler Project Manager PTV America, Inc. 1128 NE 2nd St, Suite 204 Corvallis, OR 97330 541-754-6836 x205 541-754-6837 fax www.ptvamerica.com

Not sure what's up with the mailing list here. For some reason mails are not coming through. Try again... I am trying to link an asterisk box to my provider's asterisk server via SIP. (I know I could use IAX, but the provider does not allow that, so I can't). When an inbound call happens I get this: Jan 31 13:09:14 NOTICE[3716] chan_sip.c: Failed to authenticate user

Hi, I just got my ip500 back after months of waiting. Is there an easy way to get it hooked up to asterisk without [t]ftp servers and all that or is there a quickstart page somewhere? tia r

Hi list, I was testing out the "doSMP" package from REvolutionR in my regular R2.11.0 installation and I got the following error message.? Well, one obvious thing is that R2.11.0 was built using "i386-pc-mingw32" which is different from what revoIPC used.? I could just use REvolutionR, but all my R peripherals were set up to work with the regular R2.11.0.? So, I really want

Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's

...tion to operate on a vector of 1 byte integers. So that is why I was hoping to find a way to do it with existing (compiled C code) R. Maybe I am missing something...I don't know :) Thanks for your help. Ben Stabler PTV America, Inc. 1128 NE 2nd St, Suite 204 Corvallis, OR 97330 541-754-6836 x205 541-754-6837 fax www.ptvamerica.com

...the foreign library are now used for dbf I/O, which significantly improves the speed. There are probably a few bugs in there that I did not catch, so please email me if you find them. Thanks. Ben Stabler Project Manager PTV America, Inc. 1128 NE 2nd St, Suite 204 Corvallis, OR 97330 541-754-6836 x205 541-754-6837 fax <http://www.ptvamerica.com/> www.ptvamerica.com [[alternative HTML version deleted]]

Hi I have set up an Asterisk System with One XLite Phone and when i call the trunk line or receive calls via a trunk line (FXO generic X100P) i'm getting one way Voice. I can hear the called party - but they cannot hear me... Any ideas - is t a NAT issue or is it something to do with the generic X100P card. How does one sort this problem -- Mark D'Cruz D'Cruz Consulting

I migrated to a new version of the voip connection vs1 server software and I am now getting these errors when I try to call my voicemail. Any thoughts? The files are there, so I don't get it. Oct 11 19:57:26 WARNING[6587]: format_wav.c:140 check_header: Not a wav file 49 Oct 11 19:57:26 WARNING[6587]: file.c:418 ast_filehelper: Unable to open fd on

Hi, I have "callwaiting=yes" in my zapata.conf, and "Call Waiting Serv: Yes" in the PAP2s. However, there's sitll no callwaiting on the PAP2s. Everything else work fine. Any ideas? Am I missing something somewhere? Thank you. AK -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everybody, Do you know if it's possible to push the status of an extension (a phone) to a phone like blinking a light on the phone ? And do you know wich brand of phone can do this ? I'd like to make the same as the secretary phones that can see the status of lines before putting a call on it or transfering someone to. As i know that the Flash Operator Panel get the global

I am attempting to setup a asterisk server to take place of my current service with freedomvoice. With the current system a auto-attendant picks up and they go through all the normal menu stuff, once they select the department they wish to speak to the attendant asks them to say their name. Once they do that the system attempts to contact a agent and when that agent picks up the

Hi, For thanks to everyone that answered the "dial from pph". On an other subject, how would I go about playing a wav file while talking to someone over a Zap channel ? Let me explain. I am on line with someone. I want him to hear a WAV (or mp3) sound file. I punch a key on my phone keyboard and he hears the sound file and after we can continu talking. Any hints