similar to: Cisco 7960 International

Hi- I have a 7960 using chan_sccp which gives me a busy signal as soon as I dial the 1 in my string of 915552223333. Can't figure out why. I do have a dialplan.xml file: <DIALTEMPLATE> <TEMPLATE MATCH="*" Timeout="5"/> <TEMPLATE MATCH="#..." Timeout="5"/> </DIALTEMPLATE> Anyone have any insights?

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"/> >

Is there a way to make the Cisco 7960 SIP phone dial out automatically without having to press the dial button, once the numbers that you have entered match a specific pattern? This feature is present when the phone is working with a Cisco CallManager. For example, if all of my internal extensions begin with a '5' and are four digits long, if I dialed '5123' on the phone, the

Hi all, I?ve ran into a problem regarding D-channels. I have setup Asterisk (CVS-HEAD-03/21/05-16:41:57) on RH Fedora Core 3(2.6.10-1.770_FC3) I also have a digium wcte11xp card for connetivity to the PSTN(E1). When I start zttool i see that Current Alarms changes between Recovering and Blue Alarm/Recovering. I started to see these problems after i moved the digiumcard from one PCI slot to

Hi, I have been using Cisco 7960's with Asterisk for years. I am trying get a 7961 working and have a problem. In my configuration, not all of my line appearances register to the same Asterisk SIP server. I have an Asterisk server at home and another at work. My Line 1 button registers to the home server and my Line 2 button registers to the work server. This has worked for years

Hullo :) I'm using CVS-04/23/04-23 from the stable 1.0 branch on kernel 2.6 - since I have no Digium h/w, I've just managed to get the zaprtc module to compile and run, so I thought the best way to test it would be via MoH. The MP3Player application works great .. exten => 6901,1,Answer exten => 6901,2,MP3Player(http://127.0.0.1:85/ES/28) This will play callers BBC Radio 4 from

I run a SIP-only shop with a 23 channel PRI and single T100P. Here are my configs: /etc/zaptel.conf: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us /etc/asterisk/zapata.conf [channels] context=default switchtype=dms100 signalling=pri_cpe pridialplan=unknown rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=no hidecallerid=no callwaiting=no

Hello list, does anyone know how to change the "interdigit timeout" when using Cisco IP Phone 7940/7960 with SIP-Firmware and Asterisk? it's default value is 15 sec. but i have nothing found to set this in tftp-config file etc. Thanks in advance, Roland

My first Asterisk install: Debian sarge with the 2.6 kernel, and two X-Lite soft-phones. I followed the online how-to documents and was calling between the two soft-phones and calling the demo system with no problems and had full audio. I then went on to configure the TDM400P's two FXS modules. I got into that a ways and was having some success, but no dial-tone when I was off the

continuing discussions of 79xx issues. i've seen referenced and am experiencing difficulty getting a 7970 to work behind NAT to a public asterisk server. i am successful with 7960s. 1. SIP load is 70.8-3-3SR2S 2. config works fine if 7970 is connecting to an asterisk server a local LAN (same subnet) 3. when debugging it in a NAT'd environment I see the register and

> On a related note, how do you get a Cisco 7940 to dial numbers with a > hash in them, instead of just using the hash as a dial key. For > example, I have *#21# to check diverts, but the phone will just dial > "*" as soon as you type the # after it. <DIALTEMPLATE> <TEMPLATE MATCH="#..." Timeout="5" User="Phone" /> <TEMPLATE

Hi all, could someone tell me what this does mean "bad file descriptor" when trying to start asterisk. It goes till the CLI command and then die with this message. Below an strace output from asterisk -vvvvvvvvvc It's on debian Sarge kernel 2.6.7 with packages from debian VoIP team. The server was running fine till now with this version. Thanks

> > No, this is not the case currently with any of the Cisco SIP software > loads that I am aware of. If you find this to be incorrect, please > let the list know. Cisco has not deployed much of the featureset in > their SCCP phones (such as paging/intercom) into the SIP phones due > to lack of standards/interest/political capital. > > JT Ok, after further

Hi All, Just a heads up - I was looking around the Cisco FTP a little while ago and noticed that the SIP 7.2 images for Cisco IP Phone 7940/7960 were released yesterday (16th August). No new features - all bug fixes according to the release notes. I've already started using it. I thought those of you running the Cisco phones and the appropriate access who didn't yet know would like to

Anybody know the proceedure to factory reset the a 7960 phone running 6.3 SIP software? I've tried holding # when booting the phone and nothing, i can do that on my 8.2 phone but this phone i just got with 6.3 isnt working. Also **# doesnt work either.. -- ~Shaun

The problem is some calls from the PSTN have hidden caller id so if you want to change it to something else then modify chan_sip.c #define CALLERID_UNKNOWN "Asterisk" I've changed mine to: #define CALLERID_UNKNOWN "Unknown" -----Original Message----- From: Shaun Ewing [mailto:sewing@gmail.com] Sent: 22 September 2004 14:16 To: Asterisk Users Mailing List

Typo in your OS79XX.TXT P00 ? instead of P0S !? -----Original Message----- From: Michael L?jtnant [mailto:ml@zyxel.dk] Sent: 17 August 2004 13:31 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released Hi Shaun, Saw you post, and rushed to their ftp-server and downloaded it :-) But, I can't make my phone (7940) upgrade, so maybe you

Hello all, I know this isn't strictly Asterisk, but I'm sure that there are more people here using the Cisco 7960 w/ SIP, so I thought I'd post here. I've just bought a Cisco 7960 phone to use with Asterisk. It came with the CallManager image on it. I've got the 4.4 SIP images (P0S3-04-4-00). If I put "P0S3-04-4-00" in the OS79XX.TXT file, the phone downloads this

What headsets have people found work well with the Cisco 7940 and 7960 phones? To date, I have tried a couple of the headsets within the Plantronics H series (H41-N), and noticed that the volume of my speaking is lower over the headset than on the regular handset. I am currently looking for headsets that are known to work well. I do know that Cisco lists the H-91 and H-101 as certified to

Hi, Today I was assigned a block of IPs by my ISP and decided to move the Asterisk box onto a public IP. I am also using 10.1.0.0/16 on my LAN, with the IP phones under 10.1.4.0/24 There is clear routing between the public and private IP ranges (no NAT). Private can see public and vice-versa. I changed the proxy config in the phone to point to the new IP address. Previously the 7960 worked