similar to: Using Skinny with a 7905G phone

I have a new asterisk installation (1.4.2) that is working fine with SIP. Now I'm trying to add 2 cisco ip phones (7960) running SCCP (latest chan_sccp). I have the phones booted, and the tftp directory all setup, etc. But the phones do not quite work right. When I lift the handset I only get a dial-tone 1 out of 5 or so times I try, though hitting the speaker button works. I can dial

Can anybody recommend a good web interface for asterisk that actually works. I am looking for a web interface that can show how many callers are on the phone, should be able to transfer the calls and disconnect. I have tried using the flash operator but has been unsuccessful in making it work. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com

Hello, I tried to configure a cisco 7905 IP phone using the skinny channel but I had not much luck. The relevant portion of skinny.conf is: [cisco1] device=SEP000F3487F8E3 callerid="Alex" <123-456-789> mailbox=500 callwaiting=1 transfer=1 context=default threewaycalling=1 line => 500 ; Dial(Skinny/500@cisco1) I set up the tftp server, and prepared the following

Hello! My Dom0 config: Xubuntu 8.10 + "Xen-3.3.1 and kernel-2.6.18-xen.hg" I have Nexenta CP2 RC1 (Opensolaris b104) on ZFS file system. I installed this OS in HVM. For PV i used pygrub. My config file: ''ncp2.pygrub'' bootloader = "/usr/bin/pygrub" boot = "c" memory = 512 name = "NexentaPV" uuid =

Hello As i have a java project to do i start and stop compiz with gnome-compiz-manager from gandalf and i just tried to relaunch compiz while the swing window was focused and all the interface just freezed. Worse than that it hangs all the interface and the keyboard events are not processed so you can't do anything to kill compiz and start again. But hey i have two computers so i have

Hi, I am posting my question to both storage-discuss and zfs-discuss as I am not quite sure what is causing the messages I am receiving. I have recently migrated my zfs volume from b104 to b134 and upgraded it from zfs version 14 to 22. It consist of two zvol''s ''vol01/zvol01'' and ''vol01/zvol02''. During zpool import I am getting a non-zero exit code,

Hello, I setup a NAS based on EON 0.58.9-b104 (Osol 2008/11) incl. a zfs (raid-z zpool). This is shared in my LAN via smb = working. System is running from a flash USB stick. I now installed Osol 2009/11 on a flash disk in the same machine, so I be able to boot either EON or Osol 2009/11. I tried to import the zpool in Osol, the zpool is not listed available, instead I get a ''permission

So I'm not getting caller ID via the X100P card. I've confirmed the PSTN is sending me caller id by plugging in a little third party box. Ideas, tips, greatly appreciated

I have looked all over the site(s) for help. But heres the problem. Im missing something. In coming works fine from FreeWorld via IAX. But when Dialing out i get: May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I don't know how to authenticate iaxtel to 65.39.205.121 my IAX.conf if as follows [general] port=5036 register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net

Hello everybody, I have installed the Billion ISDN on a Debian machine. I proved to call with a ISDN telephone conected to ISDN Box and it is OK. So I connect the Billion ISDN to the ISDN Box and I call from a extension to outside. But it doesn't work, that is what I have in the CLI: *CLI> -- Executing Dial("SIP/101-f9eb", "ZAP/g1/943833473|45|tTwW") in new

Hi, I've upgraded my Asterisk to 1.0.7version patched with bristuff 0.2.0-RC8c. I'm using the same extensions.conf but it seems now include instruction doesn't want to work, here follows an extract: [inbound_menu] include => ins_exts exten => _X.,1,Answer exten => _X.,2,Wait(1) exten => _X.,3,Background(msg) exten => _X.,4,Background(3-sec-pause) exten =>

If you have been paying attention, you already know this, but this weekend I have spent time ironing out the various details with my chan_skinny code that has been out there, if you knew where to look. I believe I now have all basic features operational and am going to be working on getting the class 5 (hold, transfers, call waiting and caller*id, etc) operational in the comming week(s).

Chris you might know the answer to my HUUUUUUGE problem A few weeks ago you posted this message: "I have an ISDN PRI supplied by NTL (ex Diamond Cable, Nottingham) which is currently working happily with an SDX Index phone system. I have to replace this phone system shortly and I've been trying to get a * system working for some weeks now. I have configured the dial plan (which works)

How do I make Lithuanian characters display correctly in R graphics? Instead of the special characters for Lithuanian language I get question marks... I use Ubuntu Feisty, the locale is utf-8 ... Do I need to specify somewhere the locale for R, or - default font for the graphics? -- Donatas Glodenis http://dg.lapas.info

I have a few Cisco 7905G phones and I having a little trouble configuring them. They are working with Asterisk. I'm able to get the sip image loaded, but I can't get the phones to blind transfer. Does the Cisco 7905G Phone use XML Services? If you are using the 7905G phone, would you post any of your configuration files so I can try and figure out where I'm going wrong? Thanks for

Hi All, I'm looking at getting some Cisco VoIP hardware to play with in combination with a Asterisk server. I've heard that there is beta software available to do SIP on the 7905G. So, I'm thinking of either getting a 7905G or a ATA186. My dillema is, which one to buy? I can get both for about the same price, has anyone had any experience with using a 7905G with Asterisk? On

Is the 7905G Firmware the same as the 7960 firmware? -- MBM

Hi to all, I've just received my Cisco 7905G ipphone. I want to connect it to asterisk server but it looks that it has been preloaded with sccp protocol, so I guess I need H.323 or SIP firmware image of some kind. I have a working tftp server on my asterisk box also....What do I need to do now to get things wokring? Thanx in advance, Victor...

I bought a couple of 7905G phones with a Callmanager license but i found on the site these phones can have a SIP image (which i downloaded) but before i upload the image i want to know if anybody tested them ? Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031016/4c2821af/attachment.htm

Hi, How well to the Cisco 7905G or Cisco 7912G phone work with Asterisk? Cisco claims both phones do SIP. I was strongly considering Polycom phones. However, it appears to be quite difficult to obtain support or firmware for Polycom phones. On the other hand, I find Cisco is very well supported. Thanks, Adi