similar to: Configuring cisco 7940

The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? I'm concerned as anyone that could guess an extension number&password could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there

Hi All, Having searched the archives, I can see there has been much discussion at various points regarding capture of caller id information from good old BT. If I understand correctly, it seems that not only do the drivers not currently support it, but my X101P possibly/probably can't do it anyway due to hardware? So, that leaves me with the modem route, which seems more and more unlikely,

Hi, I have access to two providers. On one of them the authuser is the same as the username, so outgoing works. On the other one I can only get incoming - what ever combination I try for outgoing I get an error. The register command has the ability to specify both usernames (which is why incoming works) but outgoing doesn't seem to, and without that I'm stuck. They are defined as:

I have a done google seaches on convertion and so far they all failed. Rich adamson and wheely-bin.co.uk Here is what I have Laptop running solarwinds tftp with the following files OS79XX.txt <- POS30201 SIP<phonemac>.cnf.xml RINGLIST.DAT <-Ringer1.pcm Ringer1.pcm The phone (7940) is hardcoded to the TFTP ip of the laptop and has a hardcoded ip is was on firmware version:

Hi Timo, I've got a weird problem on our dovecot. We're trying to setup private shared folder access. I have 2 users. tom at mailbox.com and system at mailbox.com. I have set the ACL's on all the folders (and subfolders) on system at mailbox.com. However, I'm not getting any of them listed when I check tom at mailbox.com <mailto:tom at mailbox.com's> 's IMAP

So I just saw this VoIP-centric article at slashdot (http://slashdot.org/article.pl?sid=04/05/22/1840220) which mentions e164.org. It's a "non-profit public DNS root designed to map phone numbers to Internet protocols." Is anyone on this list actually using this? They have asterisk config instructions: http://www.e164.org/config.php I wonder if someone can help me understand

Help for help says: The 'topic' argument may also be a function call, to ask for documentation on a corresponding method. See the section on method documentation. and The authors of formal ('S4') methods can provide documentation on specific methods, as well as overall documentation on the methods of a particular function. The

Hello, I am relatively new to Asterisk and I need to compile the G.723.1 codec for Asterisk. I downloaded the ITU source code, placed it in the codecs directory, but apparently Asterisk needs a rather different library than the one provided from ITU. As I've seen in the mailing list archives, there are quite a few users who were able to compile G.723.1 in *, so, could someone kindly share it

ztdummy successfully compiles under kernel 2.6, but when I load it I get ztdummy: Unknown symbol fill_td ztdummy: Unknown symbol insert_td_horizontal ztdummy: Unknown symbol uhci_devices ztdummy: Unknown symbol uhci_interrupt ztdummy: Unknown symbol alloc_td ztdummy: Unknown symbol unlink_td ztdummy: Unknown symbol delete_desc I had a quick look at the source, and it looks like these function

Hi, I deliver a test email (with 4 spaces in subject) into a maildir: Return-Path: <user at example.com> Delivered-To: <user at example.com> Received: from suse11 ([10.0.8.7]) by suse11 (Dovecot) with LMTP id Pf6zDTCaiFOUXwAA4Q5jHg for <user at example.com>; Fri, 30 May 2014 16:48:16 +0200 From: <user at example.com> To: <user at example.com>

Hi, I've been looking at the problem of the default caller ID. When a call comes in with no CID or witheld it's always set to 'asterisk' which is what the phone displays. I've been looking for an option to change that. The only place I can find is DEFAULT_CALLERID in chan_sip.c. This is set by the 'callerid' option in the sip.conf. However the documentation

On Fri, Sep 21, 2012 at 01:02:23PM +1000, John Krstev wrote: > Hi Konrad, Hey John, Please next time also include xen-devel on the To header. I''ve done that for you. > > I refer to your patch at: > http://lists.xen.org/archives/html/xen-devel/2012-01/msg01927.html > which I found reading > http://www.gossamer-threads.com/lists/xen/devel/256197 > > I have a

why can't we compile the asterisk coading in windows, it's done in c++ so it should work in windows as well

Is there a current way to specify that an instruction or function call cannot be duplicated and thus any optimizations that might want to duplicate this instruction would fail? The problem deals with barrier in OpenCL 1.0. One of the conditions of using barrier is that if a barrier exists inside of control flow, every thread in a work-group must execute the barrier instruction(6.11.9).

http://wiki.freebsd.org/JailResourceLimits Is this anthing people are working on? Is it on its way to RELENG_7? Is there a 7-version of the patch or anything? This would be a _VERY_ useful feature. -- Peter Ankerst?l peter@pean.org

On 01/05/2018 02:27, Thing wrote: > Hi, > > So is the KVM or Vmware as the host(s)?? I basically have the same setup > ie 3 x 1TB "raid1" nodes and VMs, but 1gb networking.? I do notice with > vmware using NFS disk was pretty slow (40% of a single disk) but this > was over 1gb networking which was clearly saturating.? Hence I am moving > to KVM to use glusterfs

Ok I know this may have been covered and I did have a look back in the archives but didn't find anthing so I am asking it now.. Many moons ago the MySQL CDR functions and MySQL Voicemail functions had to be removed from the main asterisk code because of licensing issues.. Now there is new MySQL stuff like MySQL FRIENDS for SIP and IAX definitions.. So how is it that these options

This is Off-Topic for Asterisk, but I wanted to get some feedback on headsets for Cisco 7940/7960 phones. We have about 10-20 people who wants/needs a headset for their phone & was hoping to collect some real-world input. Thanks!!

I am using Sipura 841 phones and Asterisk CVS-v1-0-06/14/05. Whenever a call comes in with blank CIDName or CIDNum the phone reports the respective variable as "asterisk". I can manually set the variables to whatever I want: CIDName (alpha-numeric) & CIDNum (Numeric). But if I try to make them blank, or null, or maybe throw some alpha characters into CIDNum, they get reported

Dear R Users, I suppose this is a school boy question, but here it is anyway. I'm trying to re-create the residuals for a poisson GLM with simulated data; x<-rpois(1000,5) model<-glm(x~1,poisson) my.resids<-(log(x)- summary(model)$coefficients[1]) plot(my.resids,residuals(model)) This shows that my calculated residuals (my.resids) are not the same as residuals(model). p 65 of