similar to: Tones on a Cisco 7960 ?

I have a smartnet contract for this phone, and have searched high and low for this file on the Cisco website. I need: United_States/7960-tones.xml English_United_States/7960-font.xml Every road seems to lead to the Call manager express downloads... I don't have a CME, so that's basically useles. Can anyone point me in the right direction? Mikel

Hi all, I have a working (it was a pain) set of Cisco 7960 phones. In order to dial I have to either pick up the handset or select the line and then dial the extension or outside line. How do I configure the dialplan so I can: - Start dialing via the keypad and have the phone automatically go to speaker on the first line? - Give the user dialtone after they dial '9'? A while ago I

Anyone know if it is possible to load your own XML scripts on to the phone, bypassing the Cisco CallManager? I am still waiting for my phone to arrive, but I have been playing with Cisco's phone services emulator, and that doesn't seem to like anything I pass to it. If it is possible, anyone want to share any sample scripts they have. --Mike

Hi ! On Cisco 7960 (with or without 7914 add-on module) when I press speakerphone button (or select line with line button - which automatically put second line on speakerphone) after about 15-20 seconds of dialtone Asterisk stable dies (seg fault). Tested versions of Asterisk are 1.0.2, 1.0.3 or 1.0.5, chan_sccp is newest form CVS of chann-sccp.sourceforge.net ). Firmware of 7960 is

I am trying to integrate a Nortel Norstar system with an Asterisk service using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel. The problem is when someone dials from the Nortel PBX to the Asterisk server. Asterisk answers the call and provides a dialtone (with DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only

Greetings: We have been running a CVS HEAD version of asterisk from Mar 10, 2005 on ix86 (PIII-600) Linux 2.4.27 with ixj (chan_phone) hardware. In a hope of getting better 'chan_skinny' support (to attempt using a Cisco 7920 IP phone) I built asterisk 1.2.40 on this box. Initial tests verify that our previous dialplan is working (iax2 trunks, register sip phones, registering withour SER

Hi, These days I run into something quite odd. I have an A@H that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the time. I works flawlessly with incomming SIP calls from several providers, IAX calls from FWD and with ZAP. Recently we came out with a situation where it doesn't work... with a

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 --

Hi, I was wondering if it is possible to setup with Asterisk a Cisco 7960 to use extension mobility / roaming. Meaning that a user logs into a phone and his profile moves with him / her. I have a network of ~75 Cisco 7960 phones, running SIP 7.5 distributed across 2 asterisk servers in 2 cities. I would like to enable the users to have the capability to move from one phone to another one

Hi, I see applications for signalling busy, congested, ringing, progress etc, which I understand can be provided either in or out of band. But all I want to do is generate a dialtone. This obviously can only be done in band. There is code for generating the tones when you have a physical line, like the alsa channel, or a zap channel. But I'm just thinking of if they've selected an option

Hi, My system is like this currently: ATA-186 <-> *1 <-> IAX2 to Europe <-> *2 <-> i4l <-> voicemail at cell provider When I dial up to my voicemail at my European cell phone provider I can't press '#' to get into their menu. It seems like it just ignores any DTMF tones or doesn't get them. When I call a human on the other side of the i4l they

i need to know something in the zaptel configuration as it seems i can configure detecting the busy tone and hangup after number of busy tone counts, that was great but the problem is sometimes the pstn line has no dialtone and when i try to make call it continue dialing while not having a dialtone! while it should say "all lines are busy/congested" how can i configure that?? i already

I'm sorry if this has been answered before. I seem to remember reading a similar thread a while back, but for the life of me I can't find it (after 30 minutes of intensive googling). I have a voicetronix openline 4 card in an * server running CVS HEAD as of 1st March. Everything is working pretty much as as expected (with a little more echo than zaptel?). The question I have is

On systems even key systems it is customary to have an 'internal' dial tone. Since Asterisk simply ignores the 9 and keeps the tone going it is hard to tell for some 'new users' if they can make a call. My first idea was to change the generated dial tone via source. Then if the user presses 9 go to a different context where I would record about 30 seconds of the normal dial

Greetings Asterisk users! I'm trying to setup Asterisk system to act as a callback system together with callcentric (http://callcentric.com) but it appears that I hit common DTMF issue and I want to workaround this problem. Basically my current setup is the following: 1) I have dedicated Asterisk server that it is linked to my callcentric account 2) I have US phone number (DID) from

Guys.. Im new to asterisk but is it possible to simulate a dialtone for example, in other PBX when you pick up the phone you can hear a certain dialup, which is the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is this possible? __________________________________________________________________ Anton Krall

Hello all, I have one Asterisk 1.4.21 system connected to a North American POTS line. Normally hangup detection works fine, and Asterisk hangs up properly if you are talking to a caller and they hang up; but occasionally a call comes in (typically from a US telemarketer) where the caller hangs up just as voicemail recording is starting, and you get a voicemail recording of dialtone (then

In article <Pine.LNX.4.64.0603270059460.8782@systems.shsu.edu>, amdtech@shsu.edu says... > Absolutely right :) > > "\" escapes the next character, so if you wants *69 to go through > immediately, you'd put "\*69" so that the * gets recognized as a digit. > > "," returns the dialtone sound. When my users hit "9", they like to

Hi All, I am running into a problem setting the ringtones via _ALERT_INFO on the Cisco 7960 phone. I am using * 1.2.1 and have tried setting the variable to several values. I have also tried setting the phone's software to both 7.5 and 8.2 thinking that it might be a version issue, but with no success. I have examined the packets and do see the ALERT_INFO header being sent, but the

In article <Pine.LNX.4.64.0603211635320.7043@ab1-1-246.shsu.edu>, amdtech@shsu.edu says... > You have to set up a dialplan.xml file in your tftpboot directory for the > phone to pull: > > <DIALTEMPLATE> > <TEMPLATE MATCH="9,59....." Timeout="0"/> > <TEMPLATE MATCH="9,29....." Timeout="0"/> >