similar to: Cisco ata-186 port died

Displaying 20 results from an estimated 8000 matches similar to: "Cisco ata-186 port died"

2004 Jun 12
5
MWI on Cisco ATA-186 (SIP)
I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki mailboxnumber@context ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a
2004 Jun 12
1
Cisco ATA-186 Firmware upgrade
I am currently running 2.16. Is there good reason to get the update to 3.1? Anything significant? Otherwise I am happy how it is, i just don't want to miss out on anything.
2004 Jul 31
3
MGCP & Cisco ATA 186 Help
Does anybody has the expirience configuring Asterisk with Cisco ATA 186 MGCP firmware ? I have Cisco software v3.1.1 atamgcp (Build 040629A) Asterisk 1.0-RC1 On ATA i only put domain test. mgcp.conf looks like this [test] host = 192.168.195.55 context = default line => aaln/2 line => aaln/1 Asterisk CLI shows this: Jul 31 16:05:40 WARNING[135449600]: chan_mgcp.c:485
2003 Dec 19
4
nat router + sip phone adaptor (+adsl modem)
Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for
2010 Jan 29
2
Questions about asterisk and spa2102
Hi there! First mail on the list :) 1.- is it possible to use an spa2102 to make and revice calls from a "normal" phone? I mean, I know I can use it to connect an analog to an asterisk server, but I want to know if it can be used to connect asterisk to the analog phoneline. 2.- I'm trying to unlock the spa2102 with no succes at the moment, any links or hint will be very
2008 May 27
2
GMM estimation
Hello there!!! Sorry to bother you all with such question and difficulties that I have been facing on. Recently I have been searching for packages to run GMM estimatives with R. I have been searching for such packages for a while, but since I am a new user of R system, my quest so far was unsucessful. That´s why I had decided to ask to this forum. Hope that anyone could help me! I know that
2004 Jun 14
1
making * more like a normal pbx (cisco ata-186)
I've done something similar at home, but made my dialplan such that I can dial either 10 or 11 digits locally. I don't use a "throw away" digit at all. Any 7, 10, or 11 digit call will be appropriately mangled and sent out the PSTN / VoIP provider. ________________________________ From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On
2004 Sep 20
2
Cisco 76XX - How to ignore a call (silence ring)
I am preparing to setup a system using Cisco 7940 and 7960's I have the 7.1 SIP firmware on them. One issue I have run into is how to silence the ringer if a call comes in and you don't want to take it. Many phones have a DND button. I know the 79XX has the DND in the menu but it is to cumbersome to go into the settings then phone preferences then the DND and select yes. Is there any other
2006 Mar 29
5
Problem with setting ringtones on Cisco 7960 phone.
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
2006 Jun 07
1
Controlling Cisco 7960 Ringtone from Asterisk
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
2006 Jan 18
5
SMS to fixed phone line
Telstra (Australian Telco) has recently introduced a feature to allow the sending of SMS direct to fixed analogue lines, with an appropriate handset. As best as I can figure out, this appears to use CID type signalling, or at least on a line that otherwise has no CID on it, CID is sent, but with a standard modem I can only receive the date, time, and phone number (eg normal CID info). After that
2013 Feb 12
3
improving/speeding up a very large, slow simulation
Dear R help; I'll preface this by saying that the example I've provided below is pretty long, turgid, and otherwise a deep dive into a series of functions I wrote for a simulation study. It is, however, reproducible and self-contained. I'm trying to do my first simulation study that's quite big, and so I'll say that the output of this simulation as I'd like it to be is
2004 Jan 30
2
Can Asterisk act like a normal sip phone?
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello everyone, I'm relatively new to the subject - so pleace don't punish me for idiotic questions. ;-) Can Asterisk act like a normal Sip phone and e.g. connect to another sip-gateway? Background: There is a new german company at: http://www.sipgate.de (sorry German only page) They offer a a gateway between a real telephone number and
2007 Feb 22
1
Asternic Flash Panel
Has anyone gotten this configured to show all extensions vertically instead of filling up the window. If so would you mind sharing your configuration Yes I have tried searching terms like +asternic +op_panel +vertical and a slew of others. Unsucessful though. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo echo @infiltrated|sed 's/^/sil/g;s/$/.net/g'
2008 Feb 08
1
putting mean and sd on a histogram
Hi, I would like to put the mean and + / - the standard deviation as lines on the x axis of a histogram. My attempts using the histogram function have been unable to do this. My searches are unsucessful on this subject. Any ideas are appreciated. Thanks stemp <- 5 6 5 5 5 5 6 5 m <-5.25 stanD <- 0.46291005 using windows xp R 2.6 --------------------------------------------------
2012 Nov 09
1
Fwd: Simulate nested data
I know this seems like a very easy question (and maye it is) but I've been trying to simulate nested data and been unsucessful so far.. I want to simulate a varying number of species within a group; and then create an array to store the results of my for-loop. For example: groups<-3 species$groups<-as.integer(runif(groups,1,10)) #species per functional group ###create arrays to store
2004 Jul 19
5
Cisco 7960 SIP V6 and distinctive ring.
Hi Can anyone with distinctive ring on their 7960's possibly post how they've got it to work? I understand that the ALERT_INFO variable is involved but using the examples for the variable value from the WiKi I'm just getting an error message from the Asterisk concole. Thanks in advance. P
2005 Mar 19
2
RE:Newbie question
It said 'include zapata-channels.conf', where this line wasn't commented bij the ';'... Could you post me a working example of such a config (or a part of it, for the X100P cards...? Thanks guys! Message: 9 Date: Sat, 19 Mar 2005 18:04:26 -0500 From: "Jeff Glassman" <jrglass@columbus.rr.com> Subject: [Asterisk-Users] newbie question To:
2004 Sep 27
3
CDW Part# for Cisco Software upgrade contract
The CDW part number is: 672205 And the cisco part number is: CON-SNT-CP7960 Hope this helps... In fact I think I will add it to the Wiki. ~c ------------------------------ Message: 11 Date: Mon, 27 Sep 2004 10:40:38 -0500 From: "W. Kevin Hunt" <Kevin@hbcorporate.com> Subject: RE: Cisco Downloads --> was --> Re: [Asterisk-Users] Cisco 7960 andAsterisk...not working... To:
2010 Jan 18
10
Dahdi/callerid issue
Hi All, Maybe someone knows this, im using dahdi in combination with a TDM400, where 2 analog PSTN lines are connected. The weird thing is tho that when someone calls the analog lines it goes perfectly fine, the line comes in and all works ok. Except: Sometimes the callerid from the caller is not the complete number, but only a few random numbers from that phonenumber, and sometimes its complete.