similar to: Cut off + sign in telephonenumber

Humm... Seems like my message didn't make it. Here we go again.. /Johan -------- Original Message -------- Subject: Re: [asterisk-users] ControlPlayback's options Date: Sun, 05 Jun 2011 22:19:18 +0200 From: Johan Wilfer <lists at jttech.se> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> On 2011-06-05 19:54, virendra

Hello, ctags(1) uses external application sort(1) for sorting the tags file. It calls it via system(3) function. Look at the /usr/src/usr.bin/ctags/ctags.c file, there are such lines here: if (uflag) { (void)asprintf(&cmd, "sort -o %s %s", outfile, outfile); if (cmd == NULL) err(1, "out of space"); system(cmd); free(cmd); cmd = NULL; } This code will be

Hi! Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Regards / Marcus

Hi R Users, I have produced a simulated scatter plot of y versus x tightly clustered around the 45 degree line through the origin with the following code: x <- seq(1,100) y <- x+rnorm(100,0,10) plot(x,y,col="blue") abline(0,1) Is there some way to generate a 3-dimensional analogue of this? Can I get a similar simulated scatter plot of points in 3 dimensions where the points

It's worth remembering that there are two syntactically similar but semantically different kinds of "expression" in DWARF. A DWARF expression computes a value; if the available value is a pointer, you add DW_OP_deref to express the pointed-to value. A DWARF location expression computes a location, and adds various operators to express locations that a (value) expression cannot, such

Hi List, Asterisk 's *ControlPlayback* will used for play any recorded file as an audio player. Is it possible that we can use it for multiple forward and rewind ? ex:- original: ControlPlayback(filename,skipms,ff,rew,stop,pause) expected ControlPlayback(filename,skip1,skip2,skip3,forward1,rewind1,forward2,rewind2,forward3,rewind3,stop,pause) : ----- Thanks and regards Virendra Bhati

Is it possible to dial more than one extension with a different CID to each extension? I'm thinking macros might be needed, but I don't have a good handle on macros. Is it possible? Any hints? BTW - this would be used for showing an internal extension to one phone and a PSTN accessible number to another phone. Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road

I am running Asterisk 1.16.2.13, dahdi 2.4.0 and libpri 1.4.11.4 on an HP ML110 G6 using Ubuntu Linux 10.04 LTS. I have two Digium TE121 single T1 port cards and a Digium AEX800 8-port FXS card. All PCI Express cards. Co-workers are hearing hissing sounds on some calls, and I am getting IRQ errors when running "dahdi show status". I see that sharing IRQs for Digium cards isn't

Good afternoon everybody. I first would like you to excuse me for my english. I have an issue with a SIP call ID which is not changed in the call configuration described bellow : I have an Asterisk Server A using only SIP protocol. Behind A there are 2 distant clients (using softphone X-lite) C1 and C2 and a proxy server OpenSIPS (ex OpenSER) P. The idea is that when C1 want to call C2,

Greetings- I'm about to dive into the process of virtualizing some of my Asterisk (primarily 1.4.x) infrastructure. In the past, when looking at virt solutions, the primary issue preventing me from moving was the lack of proper timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like to use either OpenVZ or KVM, but each seem to have independent "issues" that

Hello, I can do simple, "yum install asterisk18-*" and it installs Asterisk and Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and smooth. However, if I want to compile dahdi-linux on the same openvz then I get the error, *"You do not appear to have the source for the 2.6.32-4-pve kernel installed".* * * 1- Based on above error and Google search I have

Here's a weird problem that I'm not quite sure how to resolve. Zaptel 1.2.10 compiles just fine with "make", but when "make install" is run, this happens: [ `id -u` = 0 ] && /sbin/depmod -a 2.6.17-10-generic || : [ -f /etc/zaptel.conf ] || install -D -m 644 zaptel.conf.sample /etc/zaptel.conf build_tools/genmodconf linux26 "" "tor2 torisa wcusb

Hi, I try to find some information about CAS E1 signalling and how it's handled by Asterisk. My customer wants to connect to a BT ITS Netrix by CAS E1 E&M. The system is intended to take the channels and mix them (meetme / confbridge) and send the audio back mixed to each. The layout: BT ITS Netrix: CAS E1 E&M <-> MUX - WAN - MUX <-> Digium TE220, Asterisk I've

Hi, I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts. Firefox: Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't matter. WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684

Hi, I write some php code in AMI to working with asterisk command. I don't know exactly what is the different between AMI and AGI and witch one is better for my planning. Im planning to call party users that their number is is my panel on web. We have some operator and they can call party users via client softphone by clicking on their number, so they have to limited to call just listed

Hello, I have an Asus laptop, U36SD, with Optimus technology. The discrete gpu is an NVIDIA GeForce GT 520M with 1GB DDR3 VRAM Using Fedora 17; up to kernel 3.3.7-1 I was able to use bumblebee and bbswitch and then running optirun command (as I could do in F16). No more with kernel 3.4. Tried 3.4.0-1.fc17.x86_64, 3.4.2-4.fc17.x86_64 and 3.4.3-1.fc17.x86_64. I already opened a bug against F17:

Hello, I've just started to look at LDAP in IP telephony. 1. I've read parts of RFC2798 which defines inetOrgPerson class. I could find homePhone or telephoneNumber (multi-valued) attributes but nothing like phoneExtension. Did I miss something ? I not, what would you advise to store private extensions in LDAP db ? 2. Several SIP phones can query LDAP databases. Have you met any success

Thank you, Clayton. This is very helpful. We use the LLDB specific GDB remote extensions, and our debugger server supports "qRegisterInfo" package. "reg 0x3c" is the frame pointer. In the example mentioned above, we have SP = FP - 40 for current call frame. And variable "a" is stored at address (FP + -24) from asm instruction [FP + -24] = R3;; Thus we can conclude

Hi, I have this setup block: setup do session[:login] = ''jhughes'' @user = mock("user") User.stub!(:find).and_return(@user) @params = {:cn => "Bilbo Baggins", :telephoneNumber => "416-277-4418", :mail => "bilbo at baggins.com"} end And then this spec: specify "should update and save the

Sorry, this is the attachment. 2014-02-19 15:08 GMT+08:00 杨勇勇 <triple.yang at gmail.com>: > Thank you. > > Here is an example and the attchment contains extra files including object > file and executable file. > I want to print for example the value of "a", but lldb command "frame > variable a" displays "0" and so does "b", and