similar to: Any experience with Citel Link 3300 and Asterisk

Hi all, Is using a Citel gateway with Asterisk a good solution for reusing of the old Nortel digital phones? Would love to get some input from actual users. Any/all opinions welcome. robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070510/bc5fc18f/attachment.htm

Hello I Installed Asterisk on RedHat 9. I am currently try to configure minimum with two softphone talking to each other over the LAN. I am using X-Lite softphones from xten.com site. I defined 3 phones in sip.conf and also specifies in extensions.conf file. I am able to ring other computer but there is no voice exchange ( i can't hear any think except ring). Here is the portion of sip.conf

Hello I am trying to setup Asterisk on 2 servers PBX300 and PBX200. PBX300 has X100P card with 1 telephone line. PBX200 don't have any Zap device. Softphone from PBX200 can talk to softphone on PBX300 but no outgoing call from PBX200. I can call from PBX300 outside but I am unable to configure soft Phone defined in PBX200 to dial out side using PBX300 Zap devices. I am geting error

Hello I am trying to configure Asterisk on RedHat Linux 9 with one X100P one port card and one USB one port FXS card. I can modprobe wcusb but ztcfg always return ZT_CHANCONFIG failed on channel 2: No such device or address (6) error message. Also unable to config outgoing call using SIP SoftPhone. Any working examples of configuration files is highly appreciated. I mentoned followin lines

Hello I am trying to setup Asterisk on 2 servers PBX300 and PBX200. PBX300 has X100P card with 1 telephone line. PBX200 don't have any Zap device. Softphone from PBX200 can talk to softphone on PBX300 but no out going call from PBX200. I can call from PBX300 outside but I am unable to configure soft Phone defined in PBX200 to dial out side using PBX300 Zap devices. I am geting error

Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. If i pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if i dial any digit than there is no ring on Analog phone after hangup. Log's looks like this

samba@samba.org schrieb: > SAMBA Digest 1929 > > For information on unsubscribing see http://samba.org/listproc/ > Topics covered in this issue include: > > 1) samba-2.0.0beta4 > by xuwenwei@163.net > 2) Re: SAMBA digest 1924 > by Rudolf Spreckels <Rudolf.Spreckels@mh-media.de> > 3) Problems connecting over

I use res_snmp.so with asterisk do you provide mib --- David Yat Sin <davidy@sangoma.com> a ?crit : > Hi Harry, > The Sangoma Card when used for TDM Voice will work > under zaptel, so you > would need to perform the SNMP through Asterisk. > > Regards, > David Yat Sin > Sangoma Technologies > (905) 474 1990 x119 > (800) 388 2475 x199 > MSN:

Hi. I wanted to set diskless win95 but I read the list archives and I decided not to do it diskless. So I tried diskless windows but I got a problem. I do this. 1.- I did booting images with bootware proms. 2.- I managed to install DOS clients from microsoft. That was painfull because the one I downloaded from MS didn't connected to my samba server. I used a dos clients that creates a NT

i tried with the option encrpted password= yes , security = user in my smb.conf file.. but when i did following on my root prompt #killall -HUP xinetd #service smb restart #smbclient -L localhost so i get the problem session setup failed: ERRSRV - ERRbadpw (bad password name/password pair in a Tree Connect or Session Setup are invalid.) i'm not gettin why this is coming. i create the

I guess Alexandre will help you prepare this for upstream inclusion, but just want to make sure that my main point makes it across -- this is a 3K line patch. Please try to split it up into patches that add no more than 300 lines at a time, preferably fewer (but this isn't always possible). Ideally each patch should introduce one conceptual unit. Of course different people might draw the line

Does anyone know of a way to make an originate request coming over the management interface (e.g. AstTapi click-to-dial) include the relevant Alert-Info SIP headers to enable the originating phone to auto-answer? I've tried setting up a custom context (see below), but the dial plan is only entered AFTER the originating call is answered, so the SIP header is added to the terminating call leg,

All, Is there any way of telling which machine is doing what job? i.e. Domain Controller, local master, wins server etc. I have two machines and two workgroups but they both appear on the same workgroup. I think it is something to do with which machine is doing what. I think machine1 is acting as Domain Controller & Master Browser for workgroup1 and Machine2 is loosing all the browse

On Mon, Aug 11, 2014 at 3:29 AM, Deepak Rawat <deepaksingh.rawat at gmail.com> wrote: > Hi, > > I modified the query in res/res_config_odbc.c. > Original: "SELECT MAX(LENGTH(var_val)) FROM %s WHERE filename='%s'" > Modified: "SELECT MAX(LEN(var_val)) FROM %s WHERE filename='%s'" > > I rebuilt the code and installed Asterisk again.

I am testing a resource called venue in this piece of code (generated using script/rspec_resource) ==================== context "Requesting /venues using POST" do controller_name :venues setup do @mock_venue = mock(''Venue'') @mock_venue.stub!(:save).and_return(true) @mock_venue.stub!(:to_param).and_return(1) Venue.stub!(:new).and_return(@mock_venue)

https://bugzilla.netfilter.org/show_bug.cgi?id=1182 Bug ID: 1182 Summary: ipset (userspace) link failure at commit a43bccbe1ba363edab3be09cecf22be0df536555 Product: ipset Version: unspecified Hardware: x86_64 OS: Ubuntu Status: NEW Severity: normal Priority: P5

Hi, I am using file function to copy a directory & its content to few servers. I do it below way. I am not sure whether file uses sync mode or does a copy of every file over & over when something changes. Bcos when running the puppet daemon using puppetd -test it takes a while. Where can I find a list of function like file, user group etc & what args to pass file {

Dear Respected One, I am Deepak. I am currently working on SFTP application. I ve downloaded your opnessh 4.5 (latest ve rsion). I want a simple sftp library which I can link to my test application and perform ftp operati ons. Is it possible to seperate sftp out of ssh and form a sftp library ? And also please provide me the APIs in sftp which are similar to login, connect, get, put etc APIs in

I have an installation which is currenly using 14 DID Trunk Lines. I need to be able to use Caller ID information and currently it is not available on these lines. Is there another way to access this information? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040105/62559e22/attachment.htm

Hello, I am having some difficulty in getting Samba v.1.9.18p10 to authenticate using server security. I have couple of users that do not have unix accounts and those are the users that can not connect to the samba server. The message that I get in the logfile is server rejected the session. I have no problem login in to the server, but I also have a account on the unix server. Server