similar to: Cisco tftp

Thank you for all of the beta testing. New and improved graphics in this release along with drag and drop transfers and hold for all technologies. There's a screenshot on the link below. Also improved documentation so read the included README. There's also a sample xml configuration included. http://www.voip-info.org/tiki-index.php?page=Asterisk+WAMI Christian Hoffmeyer YottaDot

I am having a problem with Red Alarms on X100P cards. The most frustrating thing is I can not duplicate the alarms, therefore am not sure how to solve it. I have read after searching posts and the web that you can try different signaling methods which may help alleviate the problem. There is fxsks (Which I am currently using), fxsgs, and fxsls While reading the Digium site about these different

Here's my situation. I have two receptionists that answer incoming lines. Each has a 7960G with 5 incoming lines each. I'm trying to set this up so each line on each phone doesn't utilize call waiting. My problem seems to be that ChanisAvail(Sip/cisco1&Sip/cisco2&Sip/cisco3&Sip/cisco4&Sip/cisco5) always returns cisco1. Here are the sip.conf entries: (mind you,

> I've personally put up a $300 USD bounty on a win32 call manager - > hopefully a few others will help get the ball rolling : > http://bugs.digium.com/bug_view_page.php?bug_id=0000848 Is C# and .NET fine? This is already nearly done. I can send you binaries of a single user call manager, and the operator manager is in the pipe. Actually, I'll just post these for download.

where can I get the 2.05 firmware all i see is the 2.04 firmwares :-) also anyone got a fix for the horrible speaker phone on the 200's

Hello, I'm running * on a very basic configuration. I have a Wildcard TE405P with the first T1 connected to a PRI line and the remaining three to Adtran TA750 channel banks with FXS modules. I successfully configured everything to work with a couple of Swissvoice IP10S handsets (MGCP) and analog extensions connected to the channel banks. The problem I'm having is that when I pick up any

I am trying to setup the call manager and I configured the manager.conf file. When I try to telnet 0.0.0.0 5038 It says trying 0.0.0.0 >>>>> > Connected to localhost > Escape character is '^]'. > Asterisk Call Manager/1.0 > Then I type > Action:Login (enter) > Username:sam > Secret:sam > Then I enter twice > > I get Response: error

All, I am trying to configure hdlc support on T100P Digium card - everything seems ok... but it just does not work at all. I was able to compile all drivers, the light on the card becomes green when I plug T1 link.... but I even can't ping default router IP... there is no data coming back to me, so I am confused completely now. Also I tried both protocols - raw hdlc and cisco hdlc. The

I am trying to send calls from an AS5300 to Asterisk via e1 and I get this bit of information in place of routing information Going to extension s|1 because of Complete received Accepting call from '' to 's' on channel 1, span 1 Here are the relevant zaptel and zapata pieces. span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 signalling=pri_cpe switchtype=national

Hello all, My set up is a fractionated t1, with 1-6 voice channels, 21-24 data channels. I have a t100P installed in a amd 1500 with 512meg. Question: In order to set up the data channels, is the following correct? 1) compile new kernel with generic hdlc support 2) compile hdlc from hq.pm.waw.pl/hdlc 3) uncomment KFLAGS+=DConfig_ZAPATA_NET in zaptel makefile 4) compile zaptel, 5 compile rest

I have two grandstream budtetone-100 and cisco 7960g phones. When I talk via speaker phone on either of the phones I get a lot of echo. Any suggestions? Also how do I turn on the mark echo canceller. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, April 13,

Good Day everyone, Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by Cisco, however now the phone does not and will not read the RINGLIST.dat file. I've tried rebooting the phone, tried resetting the phone back to factory, have deleted the RINGLIST.dat file and reloaded the phone then reinstalled the RINGLIST.dat, and still the bloody phone will not read the file.

Hi All, I recently received my Cisco 7970 and have it up and running with 8.0.4 firmware, with asterisk 1.4. Seems to function pretty great so far, aside from a few issues. Here is what I have noticed so far, anybody have any fixes for these issues? 1. Contrary to the forums and lists I've been reading - the phone indeed does register. The little registration animation stays on the screen

Anyone know what the file format for the cisco 7912 RINGLIST.DAT file is? I have a RINGLIST.DAT file for a 7960 that is working fine. It's just a list of the ring name and the file name but it doesn't work for the 7912. I've read somewhere that the 7912 needs the ringlist.dat file to be in xml but I can't figure out how exactly it is constructed. Thanks. -- -------------- next

Hello: Does anyone know how to configure any of the Cisco 79XX phones to get custom ringers when using chan_sccp with Asterisk? I've currently got Asterisk's 05-24-04 CVS-HEAD and Zozo's 0.2 release of chan_sccp. I've tried using ringlist.dat, but that appears to only be for the SIP phones... Thanks for any input, Andrew

Hi all (is a bit of a Cisco question, but perhaps it's already been asked here). I'm trying to setup a TFTP server for dynamic configuration of my 7960 Cisco phones. Cisco docs suggest I download the following files from their web site: * OS79XX.TXT * SIPDefaultGeneric.cnf * SIPConfigGeneric.cnf * RINGLIST.DAT * P0S3xxyy.bin * P0S3-xx-y-zz.bin * dialplan.xml * syncinfo.xml I only

Hi , I''m getting random hangups on zap channels with long calls. It seems that the hungup happens after 10 minutes or so. AbsoluteTimeout is set to 0. Any other thing I should be configuring? Thanks! PHM

I'm trying to change the ring tone on my 7960 from the dialplan. I've tried the example on the wiki but it doesn't seem to work. Something like: exten => 3010,1,SetVar(ALERT_INFO=<Bellcore-dr1>) ; selects Ringer exten => 3010,2,Dial(SIP/3010,15) I'm not sure what the Bellcore-dr1 ringer is supposed to be. I've tried replacing ALERT_INFO with another ring tone

Hi, I'm after two very specific ringtones for the 79xx's... A dog barking, and a horse either galloping or neighing. I've tried making the sounds, but for some bizarre reason they're not working. I used to make quite a few ringtones for the 79xx's, but I seem to have forgotten how to do it! And to top things off, I can't even find the documentation on Cisco's site

Hello * community, I have 2 * boxes spanning a t1 with hdlc. I'm trying to route packets so people on a subnet seperate of the * boxes can browse the boxes on the * spanned subnet. The * boxes and the boxes on subnet controlled by * can see everything. The boxes on the seperate subnet inside the company can only see so far as the first * box and can't ping across the span.