search for: sbesch

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 coul...

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

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 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

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

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

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

...lved it? search THIS list to find that you must be registered to nickserv to get in the channel. It has been discussed many times and at length. -- Steven Critchfield <critch@basesys.com> --__--__-- Message: 11 To: asterisk-users@lists.digium.com From: "Stephen R. Besch" <sbesch@acsu.buffalo.edu> Date: Thu, 15 Jul 2004 16:31:47 -0400 Subject: [Asterisk-Users] Re: Updated Grandstream configurator Reply-To: asterisk-users@lists.digium.com Steve wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On Thursday 08 July 2004 05:45 pm, Stephen R. Besch w...

Hello, I have a problem with asterisk and Grandstream BudgeTone-100. With default configuration everything works (in anonymous mode and fixed IP), but if Im trying to enable registering, it dos not work. I used 'sip debug' and verbose level 10, nothing happens if I switch telephone on (no messages about bad auth etc). As I understood, after switching phone on at first it will try to

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

...lean, make opt 3. /asteriks/channels/h323, make clean, make install, and it is got error about no chan_h323.o exists. and the make install is failed. any one can help on this. Thanks, George Lin --__--__-- Message: 6 Date: Thu, 06 Nov 2003 11:32:27 -0500 From: "Stephen R. Besch" <sbesch@acsu.buffalo.edu> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] The Minimum Cost of Setting up an Asterisk Phone System? Reply-To: asterisk-users@lists.digium.com >Here's a cost analysis, rather crude and inspecific, of using Asterisk >to implement a phone system...

Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and when

From what I understand, shared interrupts are permitted on the PCI bus. Whether they work for a specific card or not depends entirely on how the driver is written. Someone at Digium should know this. -- Stephen R. Besch, Ph.D. SachsLab 320 Cary Hall SUNY at Buffalo Buffalo, NY 14214 (716) 829-3289 x106

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

Here is some more stuff to add to the confusion about the "callprogress" option. I currently have my * system operating with a T100P talking to an ADTRAN TSU600 channel bank with 8 FXO ports connecting to the outside world and Grandstream SIP phones as handset extensions. At first I naively set "callprogress=yes" in zapata.conf. The results were typical of what many

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

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 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

Several bugs were reported in the first release version, which are now fixed: 1) The file generator was losing the filename and not successfully generating files when using the MAC address. 2) The path initialization code was not working correctly. Also, I have appended the version number to the main window's title and added an option to show phone passwords in plain-text for those