similar to: How many X100P's in a system.

I have solved the time server problem with the Grandstream by having my * box's NTP service mirror a public NTP server. I had to do this because my phones are all on the 192.168 subnet, which is non-routable. For example, assuming that the NTP service is configured and running on your * box, create an NTP mirror which allows access from machines on 192.168.10.X by adding the following

I have found that the NTP server is not contacted when the phone (Budgetone 100) comes back from a power down. I must reboot the phone without powering down to get the phone to contact the NTP server for the time. It doesn't matter whether I reboot from the phone's web site or using the menu reset function: either works. I have only tested this with a private NTP server. This may

Hi Just recieved the above phone Does anyone have sip.conf and extension.conf example for the SIP phone working with the FXS w100p and the FXO tdm400d any help would be appreciated Thanks Robb

Hi All I have 3 GrandStream BudgeTone-100's which connect to an * box with a HFC-S based ISDN card using ISDN4Linux. I have setup the BudgeTone-100's to use Early Dial which for calling between the three phones works well, but for the external calls using the following extension exten => _9.,1,Dial(Modem/g1:${EXTEN:1}) Only the first digit is dial on the ISDN Line. Does anyone know of

I know the recommendation it to not run more than 2 T/E100P's in a system but what about X100P's.. Usually there are 5 PCI slots in a system, has anyone tried 5 x X100P's in a system? Later.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

The most recent version of GSConfigure is available at www.buffalo.edu/~sbesch Several serious bugs that kept the program from getting started have been ferreted out and corrected with the help of Bruce Komito. The program is now actually running on someone's machine other than mine. I have built this version with the oldest copies of the system dll's that I could find inn an effort

I've just finished a general purpose configuration utility for the GS phones: 1) Generates files from scratch (using MAC), from HTML config listing, or by directly downloading from the phone. 2) Does multiple simulteneous edits. 3) Can reboot as many or as few phones at a time as you like. I would like to offer it to the list, but there are 2 issues: 1) I want to GPL it first, if

I have Asterisk configured to record calls. Both in and out record ok but SoxMix fails to join the two files. The error from the CLI is as follows: Execute of ( nice -n 19 soxmix /var/spool/asterisk/monitor/Support-in.wav /var/spool/asterisk/monitor/Support-out.wav /var/spool/asterisk/monitor/Support.wav && rm -f /var/spool/asterisk/monitor/Support-* ) & failed. If I run exactly the

Three days after launching our * system with 20 GS phones, I have finally had to give up on dynamic registration. The phones keep dissappearing from the sip peers list, even if just sitting idle. Either I spend half my time re-booting phones to get them registered, or the extension appears busy to outside callers and people get really irritated. Even setting the registration interval to 5

Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but

hi all i upgrade a bt100 phone and it can't resgister with asterisk Dec 1 13:25:49 NOTICE[1112980400]: chan_sip.c:7519 handle_request: Registration from '<sip:@172.16.4.249>' failed for '172.16.4.226' is was working with the version 1.0.5.3 some bady now what is hapening? thanks in advance Rodney

Hi, I'm new to R, with some experience with Matlab and SPSS. I've figured out how to run my repeated measures anova and am getting the right numbers for my effects (comparing with results from other software), but am having trouble with the model.tables function. Specifically, using: model.tables(fm,"means",se=TRUE) prints the means, but then won't do the SE values,

Hello, Did anybody successfully tried upgrade Adtran TSU 600 to firmware which is working properly with T100P and asterisk ? B.

Hello, I would like to access my data frame without one variable. E.g.: > colnames(x) [1] "Besch" "Ang.m" "Arb.m" "i10" "Umsatz" "arbstd" I can try x[,-1], but this variable must be called by it??s name. x[,-"Besch"] x[,!"Besch"] attach(x) x[-Besch] ... ... does not work. I could not found a solution of

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

hello, I believe I have started guest VM now. Is there any GUI support for guest VM? I tried to "apt-get install gnome", but it doesnot work. My host OS is ubuntu 10.10. 2011/1/18 Ian Tobin <itobin@tidyhosts.com> > Hi, > > > > On debian you need to create or open > > > > /etc/modprobe.d/local-loop > > > > and add > > > > options

Does anyone out there know if it is possible to discover the dialed number when a line in an analog hunt group rings? I can't get a straight answer from our IT folks. We have a 5ess switch delivering 4 analog lines which are in a simple hunt group servicing our lab. I would like to have a different call attendant based on which number is dialed so that I can route the calls to the

This message is in MIME format. --=_6vx1wr5xhvr4 Content-Type: text/plain; charset=ISO-8859-1; DelSp="Yes"; format="flowed" Content-Disposition: inline Content-Transfer-Encoding: quoted-printable Dear R Core Team, Help to 'read.table' claims that 'read.csv' is identical to 'read.table' except for the defaults. However, 'read.table' seems

If you wan't to create a ringtone with makering.pl for firmware 1.0.50, be sure to create it as ring.bin and then rename it to ring1.bin / ring2.bin or ring3.bin. This seems to be the only change between the format from 1.0.4.68. Regards, Maron

We are in the process of migrating to new database servers. Our current RAC clusters are running OCFS2 1.2.9 on RHEL 4. Our new servers are running OCFS2 1.6 OEL5. If possible, we would like to minimize the amount of data that needs to move as we migrate to the new servers. We have the following questions: 1. Can we mount an existing OCFS2 1.2 volume on a servers running OCFS2 1.6?