similar to: Cisco 7960 Won'

Hello, I have tried to install this phone for hours now and I can't get it working. Maybe someone can help me :) I have searched for more info from everywhere but there isn't much about 7910 :( >From the CLI I get this: NAME ADDRESS MAC Reg. State ================ =============== ================ ========== telefon --

I'm new to Asterisk, but, Managed to get it working for outound calls from my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In fact, voice mail won't even work.. This is a snippet of what I'm getting when I try to call the ATA -- Executing

I'm trying to setup a couple of Cisco 7960's in asterisk. I have asterisk working fine for sip clients, and can call the 7960's just fine, but I can't seem to dial out on them. As soon as I enter the first digit, the phone attempts to dial it without waiting for the rest. I've changed timeout settings, etc but can't seem to get it to work. Any ideas? Asterisk

I've done the unspeakable. I took a working setup of 1.0.5 and upgraded with the latest cvs. with 1.0.5 distinctive ring worked great. with the latest cvs, it doesn't seem to work with my sipura 2000 (the only thing i have to test it with). I can see in console that its sending the info, but its not working? i realize the likely hood is I messed something up somehow during checkout or

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

Hey all, Anybody been successful running DHCPD on a GRE tunnel? When I tell DHCPD to listen on cisco1 I see this in the log Jul 23 16:21:03 atlantis dhcpd: cisco1: unknown hardware address type 778 Here is the output of ifconfig cisco1 Link encap:UNSPEC HWaddr 8B-8E-28-32-00-00-00-00-00-00-00-00-00-00-00-00 inet addr:10.199.0.2 P-t-P:10.199.0.2

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

----- Original Message ----- From: allan d. go To: blue@bluepoint.com.ph Sent: Tuesday, October 29, 2002 8:54 PM Subject: Samba PDC guys, i'm configuring a Samba Primary Domain Controller. i compiled the 2.2.5 version with ./configure --prefix=/usr/samba --with-quotas and followed the Samba-HOWTO-Collection I configured my Win2K workstation with the following parameters:

Hi, I have an FWD account and it's configured in asterisk. I can be called by people using FWD, but I cannot make FWD calls myself. Every number dialed with a 8 prefix goes to FWD, if for example I call the echo servie I get this: Connected to Asterisk 1.2.13 currently running on asterisk (pid = 2865) Verbosity is at least 35 -- Executing SetCallerID("SIP/timothy-08224f08",

I was wondering the same thing. The traceroute seems to end at their border router, which leads me to believe that they has a hardware failure. Hoping that they get it up in a few days. I've checked around and there are a few mirrors but they all just hang, so I guess we have to wait... Mark Fowle ------------------------------------------------------- Navicom Inc. One Steelcase Road W

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,

Hi zozo, sorry to bother you; I've been trying to get a cisco 7920 phone (that's the cute wifi/wireless modell) to work with asterisk (no success so far) and ran across your chan_sccp module; is there a current cvs version that I could base my tests on? tar version 0.1 seems to be ~2003-09-12, cvs access as posted on http://theo.me.uk/pages.shtml?page=sccp doesn't seem to work any

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

I am running asterisk 1.0.1 with OH323 compiled in. We have a 323 trunk to CallManager with a mgcp controlled pri router. When using sip phones (directly registered with asterisk) to call out the 323 trunnk to PSTN, calls timeout after 3 rings. If I answer b4 3 rings - no problem, otherwise I get "no one is available to answer at this time" on the consoel and it redirects to an

Can someone provide example configuration files and/or instructions for configuring Asterisk to work with chan_sccp2 and a 7960. I have the searched the chan_sccp2 site, the voip_info wiki site, asterisk doc project, and mailing list archives for the past year for help, but failed to find anything helpful. Some of the questions I have are: 1. What settings need to be made on the phone? 2.

Dear Skinny/SCCP lovers :-) I've just completed & uploaded to the cvs the newest version with fixed redial key AND implementation of speed dials. please test extensively and report any bugs. i know that the display is not yet set correctly but the buttons are working as expected. Enjoy testing... --jan (*1) http://chan-sscp.sf.net (*2) yes, bugtracker is down at the moment, will fix

I have a Cisco 7960 running 7.2 of their SCCP image; I am running Asterisk 1.0.5 and using the latest Sourceforge version of SCCP2. When I make a call (or receive one) the "Transfer" softkey does not show up - as a matter of fact only 2 softkeys show up (redial & something else), but those even are not active. On a 7960 running SIP the Transfer and other buttons do show up and are

Hi I have 2 asterisk servers running 1.2.12.1 and IAX2 with trunking and no jitterbuffer. Both servers are using sccp2 with 7940's and 7960's with 7914. Server 1 is my main VOIP server and is connected to the pstn and VOIP wholesale provider. Sever 2 is a branch site and all calls go to server 1. If I make 1 call on server 2 everything is fine. If I make a 2nd call so there a two calls

Hi, I don't know how to call this functionality, but what I want to do is join an already established communication between PSTN---FXS_connected_phone using my SIP phone (I have an asterisk v11 with digium TDM400P at home) Is it possible? What I don't want is using the conference sound and menu.... It's just a normal call between to channels that I have to join for few minutes.

I could use some skinny/Cisco help here. Was finally able to get the phone registered to * but whenever someone tries to call that phone it freezes and reboots itself. Same thing happens when you pick the handset up off the 7960G; it locks and reboots about 5 sec later. Here is what * shows when I plug the phone in: -- Starting Skinny session from 64.72.107.1 Device SEP000F3442E4A7 is