similar to: *78 *72 and sip?

Someone recently posted the following list as functions built into * *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid I'm running a CVS from a couple of weeks ago with multiple C7960's, snom 200,

Been learning * now for a couple of weeks and have all basic features running including VM, MoH, FX lines, iaxtel, and FWD. However, I seem to be lacking documentation on a lot of technical things and am wondering if I overlooked something that is obvious to others. (I do have the Handbook, have been doing a fair amount of google searches, and read the README.* files.) Examples, Where should I

I need to find a way to have the Polycom phones automatically park calls. Right now my users hit #70# (I know the last # is optional but it speeds it up.) to park a call. Personally I think this is easy, but my users would like one button to do this for them. The Polycom has buttons we don't need (Transfer & Services), it would be nice if I could remap one of those buttons to dial

Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein

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.

If someone could point me in the right direction I would much appreciate it. Here is my problem: My directions for my sip phone says to dial an ip address 12*34*65*78#. When I dial that into my phone my asterisk server is only picking up some of the numbers in the above example it would pick up 6578. Then of course not find it and ring busy on the phone. The same is true for dialing a regular

Hello to all I'm new in the list. My name is German. I look very pretty asterisk software, but I don't know if it have any X or web tool to configure or admin. thank to all Best regards German Aracil Boned

Hi all, does anybody know how to make on-demand anonymous calls? I've tried code *67# before the number to call but it is working with some providers only. Any hints? Thank you.

Hi what can be wrong with * that console does not show any stderr text printed from agi script? I am starting with asterisk -vvvvvvvvvvvvvvvvvvvvvvvvrc VERBOSE command does show text on console but printing of STDERR does not I tried it from Perl and from Java and in both cases almost the same result, except in Java more things do not work. In Java for, for example, SAY DIGITS 123 78# would

Hello, I''m a newbie on the rails environment. I''ve just created an application.html.erb to have the same layout on all pages but the site will be available in 2 different languages, then with 2 different layouts. Then I would like to have an application.html.erb with the layout in French and an application.html.erb with the layout in English. Thanks you for your help, --

I have seen some talk about this but have not seen any answers. I know this is a problem on CentOS 7.1 and I also think it is a problem on CentOS 7.0. Basically if I have an NFS client only config - meaning that the nfs-server.service is not enabled then I have to wait 60 seconds after boot for the 1st NFSV3 mount to succeed. What I can surmise is the following. I attempt to perform a NFS

> > 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

Hello, I am installing Puppet Enterprise agent (puppet-enterprise-1.2.1- solaris-10-sparc) on a Solaris 10 64bit SPARC system. When I view the logs for the agent I see: puppet-agent[14680]: [ID 702911 daemon.error] Could not run Puppet configuration client: Could not retrieve local facts: private method `split'' called for nil:NilClass The client successfully initiates a cert exchange

Sipura has implemented auto-answer in version 0.9.5 of the SPA-841 firmware. However, it is implemented via the Call-Info header, which Asterisk stable doesn't currently support. The attached patch implments a quick hack to support the Call-Info header from the Dial() application by way of setting the CALL_INFO variable. For example, the following macro can be used to dial up a single

Hi, I've just found: http://lists.digium.com/pipermail/asterisk-users/2003-June/014475.html which talks about ALERT_INFO and Cisco phones. How do I actually get this working and what does it do? Do I need to add anything to the configs for the phone or is it just a SetVar(ALERT_INFO=1) - which I tried and it seemed to do nothing at all.. Thanks Andy

Hi, Looking through the source code, I see two functions in the 'libpri' code for message-waiting: pri_mwi_activate, and pri_mwi_deactivate, but they are not used by anything anywhere. Is there a sample application somewhere which makes use of these functions? I believe that the ISDN MWI is used when a Norstar MICS has an external voicemail provider. I don't know exactly how this

I'd like to survey those on-list who actually use wifi SIP handsets. What type of wifi access point do you use? Are you happy with it? I presently use some older Linksys WAP54G APs. I'd like to replace these but in doing so I'd like to be moving in a VOIP friendly direction. I've yet to find a handset that I'd buy in quantity, but my last round of access points lasted >4

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Hei! I have a little problem with the subject. I use Asterisk CVS-HEAD-09/06/04-12:42:56 as a production *, but I do tests with a newer version Asterisk CVS-HEAD-11/18/04-10:01:32. Ok the problem is: in extension.conf I use macro for redirection, found on wiki pages: [macro-stdexten] ; ; Standard extension macro (with call forwarding): ; ${ARG1} - Extension(we could have used ${MACRO_EXTEN}

I am having trouble getting park to work with SIP, I have these config files: /etc/asterisk/parking.conf [general] parkext => 8540 parkpos => 8541-8555 context => parkedcalls parkingtime => 45 /etc/asterisk/extensions.conf include => parkedcalls include => default [default] exten => 3874,1,Dial(SIP/3874|20|tT) Do I need something else somewhere? Is anyone using park and