similar to: IAX2 Trunking help!

Good day all I've tried my iax conf and I'm struggling.So I want to know If someone else got this working and if they can pleas send my their configs I have to asterisk server,in different tows,both offices connected wit a direct line so both servers are on the same network running SIP.Each town got different extension register to each sever.Town A=100+ town B=200+ How do I get town A

Bear with me this is a bit long winded. I am having some issues making automated outbound calls over Broadvoice from my Asterisk 1.4.2 server. For reference, none of the below issues happen when I make the calls to VoIP phones attached to the Asterisk server. What I am trying to do is call, using a .call file, out via the SIP trunk we have setup, and when the party picks up use AMD to

Hello all. I'm having an odd problem getting * and sipgate to work together. From Sipgate support I have gotten this repsonse to my query: ===== Your Asterisk is registering incorrectly with our servers. It registers like this: sip:s@217.XXX.XXX.XXX:5076 The "s" should be your SIP ID. Anything else is rejected. I don't know where you can find this setting, but from our

Hello. My setup: _ one Samba 3.5 domain (XXXXXXXX), with a PDC and a BDC, both running FreeBSD; _ one AD domain (YYYYYYYY) running on two Windows 2003 DCs; _ bidirectional trust between the two domains. Everything used to work until I moved the PDC from Samba 3.5 (EOL'ed) to 3.6; now, users from domain YYYYYYYY cannot access the PDC's shares. I used to have in smb.conf: >

Hello All, I am having a problem with samba's ability to interpret certain files with windows 2k clients. Setup Details: Linux ( RedHat 7.3 ) File Server Win 2k Clients Samba v2.2.3a Kernel 2.4.18 statically compiled. Problem Description: When copying jpegs from a Win2k client which is displaying the JPEGS in thumbnail mode, to the Linux / Samba file server, 3 files are copied as

Dear R-WinEdit users, I got a simple question, but somehow I cannot find the answer, although I have tried a lot! I got an ASCII-file and I want to import it into R, so that every character is defined by [i;j]. The rows are not of the same length. the file looks like the following shortened abstract example: name: xxxxx xxxx age: 9.9.99 record number: 999 title: xxxxx xxxx xxx keywords: xxx xx

Hi Guys, We have just set up Asterisk (CVS Head) for a realtime enviorment using MySQL & Asterisk Addons. I have populated the "sip_buddies" table with the same information that is came from our sip.conf, however registration seems to fail for the softphone we have set up. Does anyone have any idea as to what I should be looking for here? I'm not getting any error messages

hello list, i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06xxxxxxxx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten => s,1,Ringing() exten => s,n,Playback(hello-world) exten => s,n,Dial(SIP/105) exten => s,n,Hangup() it works with one number how can i do in order to create a

Hi all, In order to do loadbalancing between my two *, i wanted to stock all things concerning voicemail on a NFS partition... I see that the voicemail system put his files onto two differents directories : /var/spool/asterisk/voicemail/mycontext etc. and /var/lib/asterisk/voicemail/mycontext etc. I've two questions : Why ? and how can i do to centralize the destination of the messages AND

Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin

Hello, I am trying to setup my Asterisk box to accept calls from the FWD network. I've followed all the config advice / samples I've found on the web. Making calls to devices on the FWD network from my Asterisk box works flawlessly, but whenever I try to call my Asterisk box from a FWD client I get a busy signal, and a "Call Disconnected" 486 error. What's odd is that I

Using CVS from 7/12/04 and trying to get the voicemail broadcast feature to work. Voicemail.conf has [mycontext] 3722 => 1234,BroadCast Test,,,cc=*@mycontext . then many other voicemail boxes. ----- whenever I leave voicemail at box 3722, only box 3722 gets the voicemail. It is not expanding it to other voicemail boxes in the [mycontext] context. Even if I replace the cc= line with

Dear all, For two sets of random variables, say, x <- rnorm(1000, 10, 10) and y <- rnorm(1000. 3, 20). Is there any way to overlay the histograms (and density curves) of x and y on the plot of y vs. x? The histogram of x is on the x axis and that of y is on the y axis. The density curve here is to approximate the shape of the distribution and does not have to have area 1. Thank you

Hi folks Is it possible to turn off the native bridging on Asterisk? I've been hacking about app_disa.c to support account & pin numbers, that tag the calls depending on who logs in..... It all works fine, then dials the destination number they requested. My setup is as follows [ENDPOINT] <IAX1> [MYASTERISKBOX] < IAX1 > [TELCOBOX]<>(PSTN) If i dial

Hello, How much would you charge, got a contact email I can get you on. Thanks Hannah Sent from my iPhone > On 15 Nov 2015, at 04:54, Scott Winterstein <0turn1 at gmail.com> wrote: > > I can help... is this a free need or can you pay a little for the programming... im good and Im cheap > > Sehr geehrte Damen und Herren, > > Hallo, > > > Vielen Dank

Also, it is possible to change "ylim" also? 2012/7/1 li li <hannah.hlx@gmail.com> > Dear all, > I have a few questions regarding the boxplot output from the > "geom_boxplot" function. > Attached is the output I get. Below are my questions: > > 1. How can I define the xlab and ylab myself? > Also I would like to remove

Hi! I have to Set up an asterisk Server with a Diva Server PRI E1-30M. Capi, asterisk, etc. everythink works. my problem is the handling of the MSN's. say, we have the block (without area-code..) 4321-0 to 4321-4999 between this numbers (including em) every MSN is possible. do I have to add all MSN's i need (several hundrets) to the capi.conf? then the routing to SIP-Phones shall be

Hi, I'm running our local school's mail server on CentOS 7, Postfix and Dovecot. We get quite a lot of spam, so I have the following sender restrictions in my /etc/postfix/main.cf: --8<------------------------------------------------------ # Restrictions SMTP smtpd_helo_restrictions = reject_unknown_helo_hostname smtpd_sender_restrictions = reject_unknown_sender_domain,

Hello, Im new to ice cast and was wondering how I would go about setting it up to allow multiple DJ's for different locations.I would like to have an auto dj playing and then Dj's want to broastcast the auto dj stops, and when the dos stop the auto dj kicks back in straight away. If anyone has any idea on how to do this that would be great! Thanks Hannah -------------- next

Is there an easy way to make graphs for the following data. I have pretest and posttest scores for men and women. I would like to form a 'titlted segment' plot for the data. That is, make segments joining the scores, with different types of segments for men and women. Example data: menpre <- c(43,42,26,39,60,60,46) menpost <- c(40,41,36,42,54,58,43) womenpre <-