similar to: Initial Audio Cut off

How can we set the CDR Master file to rollover at say 30 Meg and create a new one -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/2e98385f/attachment.htm

We are running asteriskNow 1.4.18 and after a few days it becomes unresponsive and inbound INVITEs timeout. We just reboot the box to resolve it. But it seems to be occurring more regularly now. I am hesitant to move to latest version, but will do if needed. Any guidance or troubleshooting modes I may use will be helpful. -------------- next part -------------- An HTML attachment was

Does anyone have any feelings one way or the other about running Asterisk on AMD vs running Asterisk on Intel? Thanks, Lyle J. McKarns ------------------------------------------- Networking/Linux Engineering Team n|m Nexus Management 4 Industrial Parkway Suite 101 Brunswick, Maine 04011 Tel (USA) : 1 207 319 1105 Tel (UK) : 0207 100 4968 Fax : 1 207 725 8552 Nexus Management,

Hi My call flow is T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38

We have multiple entries like the one below in our users.conf file... where the username. Contact and secret changes for different customers and we register on their behalf to the Service Provider. For the trunk below: when the call is placed out, Asterisk (1.4.18) sends the username of "abc.com" in the MD5 Auth .....which obviously does not match the trunk setup for this Customer with

In our users.map file we have the following entries: unix1 = nt1 unix2 = nt2 unix3 = @unixgroup3 unix4 = @unixgroup4 When we create shares on the samba server, we assign the valid users as @unixgroup3 or @unixgroup4. For whatever reason, if a share has @unixgroup4 as the valid users entry, we cannot connect to it from the nt side. If we flip-flop the group entries in the

Setup is: Asterisk --->NAT--> SIP Proxy I have following entry for SIP Proxy in sip.conf [Proxy] type=peer host=Static IP (NAT Firewalls public IP) username=xxxx secret=xxxxx nat=yes???????????????? canreinvite=no???????? qualify=yes Proxy sends a call and I get this error Found no matching peer or user for <NAT's Public IP:70001 NAT is using 70001 as the source port in the

Hi All, I'm having a bit of trouble with my new *NOW setup. I've downloaded and installed *NOW 1.5. We're using 1 SIP Trunk from SimpleSignal.com. Outbound calling works great, but I'm having some trouble with inbound calls. First, we would get the "the number you have dialed is not in service" error on inbound calls. After some googling, I found out that I needed

Dear all, I am trying to upgrade R on Debian 3.16.0-4-amd64, as the default R version is 3.1.1 (2014). When I try to run ./configure on R-3.3.3, I get an error message saying ... checking for curl-config... /usr/local/bin/curl-config checking libcurl version ... 7.53.1 checking curl/curl.h usability... yes checking curl/curl.h presence... yes checking for curl/curl.h... yes checking if libcurl

Hi guys, a little OT but I figured this is the place that would know. Is there a free or paid webapp where I can get inbound sms messages? I only need to receive a few inbound sms messages a month but it cant be my current cell number :-( Any thoughts? Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear Dirk, Do you mean binaries for R-3.3.3? I could not find any link on the page you mention I downloaded R-3.3.3.tar.gz from https://cran.r-project.org/src/base/R-3/ then ran ./configure I also tried ./configure --enable-R-shlib based on a suggestion found in http://jenzopr.github.io/bioinformatics/2016/05/03/r-wheezy-build.html Many thanks, Lucio On 4 April 2017 at 17:17, Dirk

My google search for Plackett's Algorithm didn't return too much except that Plackett's algorithm appears to be useful in Control Theory - it is elaborated as "Plackett's algorithm for on-line recursive least squares estimation". Sounds something like what I want. I am looking at developing a user modelling type app (new data points coming in and wanting to dynamically

My sip show channels shows some channels active that I can not make sense out of, and they have been that way for days, so I am pretty sure they are orphans. Is there a way to show active CALLS (instead of channels) to try and determine the source? Does the output below provide any clues as to why these channels might show active? Anyone aware of related bugs? The #'s indicate original

I'm interfacing to C code that uses 1-based indexing on arrays -- it ignores the zeroth element. Thus, input vectors from R must be moved up one, and output arrays must be moved down one. What is the best way to deal with this using R internal code? My current approach is: For an input R vector of length n, allocate a new vector(v) of length n+1 and copy input into v[1] to v[1+n]. Call

Hello, Suppose customer A calls attendant. CallerID of A is displayed at the attendant. But, when attendant does a consulted transfer to, let's say, B, the callerID of attendant is displayed at B. When the consulted transfer is succesful, the callerid of attendant is STILL displayed at B. Is it possible to, after a successful transfer change the callerid of the attendant in the callerid of

Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is dialled I'd like processing to return to the attendant so the user can place a subsequent call. I have setup features.conf to include: [featuremap] disconnect => ** My Dial command looks like this:

On Tue, 2016-04-05 at 11:17 +0200, Maikel van Leeuwen wrote: > Techopedia explains Production Server > > A production server is the core server on which any website or Web > application is being hosted and accessed by users. It is part of the > entire software and application development environment. Typically, the > production server environment, hardware and software

I'm having trouble finding documentation for the auto attendant does anyone have an idea where there might be some???

On Tue, 2016-04-05 at 08:16 -0700, Akemi Yagi wrote: > On Tue, Apr 5, 2016 at 8:08 AM, Always Learning <centos at u64.u22.net> wrote: > > > > > What matters for the 'free' Red Hat software is ***ONLY*** Red Hat's > > stated terms and conditions - definitely not what someone else has > > put on a web site. > Here is the link: > >

Greetings, I'm using asterisk to connect our three locations together with a sort of inter-company auto attendant connected like this: PBX (fxs) <-> Sipura 3k (fxo) <-> Asterisk <-IAX-> remote asterisk It works like this: Person picks up their phone and dials a number to get to the auto attendant (I don't have any FXO ports available on our PBX to do it the