similar to: Updated version of Grandstream cfg file generator

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

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

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 needed to create config files for downloading to Grandstream devices and made a little script for it. It seems to work fine for the HT486. The script needs a config file (cfg.in) which is in this format: P2 = blah P10 = hrm ...etc... The configfile may contain comments (starting with '#') and empty lines. Mind that the 'gnkey=0b82' shouldn't be in the configfile, as

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

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

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

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

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

Hello! I've been reading through the archives on this list for the last 8-10 months. There are some reports on success with tftp autoconfiguration with a given cfg.txt format but really vague. Has anybody successfully done this without using GAPS, or has anybody got a correctly formatted cfg.txt file that works (from GAPS). I would be happy to write a script or a java program that

I have just successfully got the TFTP file remapping to work such that I can have unique ringtone files for each and every extension. I added the following to my server_args line in the xinetd configuration for TFTP: -m /home/asterisk/grandstream/ringmap.cfg Now the entire line reads: server_args = -v -s /home/asterisk/grandstream -u asterisk -m /home/asterisk/grandstream/ringmap.cfg (There

I don't know how I managed to mess up sending this last time, but somehow it got attached to the AgentCallbackLogin thread. Since the indended audience may not see it there, please indulge me by tolerating this second copy: Here's a little tidbit about the non-functional flash key on the Budgetone 100's. I have 20 of these phones. On some, the flash key works, and on some it

I know that this issue has been discussed a lot on this list in regard to some of the recent CVS's. However, it has come up as an issue on an older release (CVS Aug 05, 2003) as well. I thought that a heads up was in keeping with the philosophy of the list. Here are the details: Call from GS via * to remote IAX to PSTN. Sound stream is established from PSTN to GS but no sound from GS

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

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

Hi list, I'm trying to figure out the format of the binary data in cfg.txt - so i'm looking for someone with a GS phone/adapter from sipphone.com (bought there so it downloads the config there also). I suppose they use GAPS there and also download the cfg.txt configuration there? Please send tcpdump data of the tftp session - or other udp dump data. I have a tftpd server hacked and ready

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:

I'm using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case) Aborted 192.168.16.32 C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.20 25 Jan 18:25 Error Aborted 192.168.16.32 C:\Program Files\TFTP

Am I the only one that has noticed there is no way to wallmount a Grandstream phone? There are screw notches on the back, but no hook to hold the handset in. -- Dave Weis "I believe there are more instances of the abridgment djweis@sjdjweis.com of the freedom of the people by gradual and silent encroachments of those in power than by violent

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