similar to: Transfer of Calls Between Legacy PBX and Asterisk

Andrei, Do you have X-Windows running on the linux box? I had a similar issue that was eliminated when I stopped this process and samba from running. Now samba is only allowed to come up during non-business hours, for changing BG music. Also, make sure your registration period in either (polycom) ipmd.cfg or sip.cfg is set to be at least the default 3600 time period. I also removed the

Hello Everybody I am using Trixbox 2.4 (with TE420P & PRI lines) .. my setup is like Calls -->Asterisk-->legacy pbx--->analog extensions(agents). Whenver a call comes in , asterisk dials the ACD number of the legacy pbx which in turn decides to route to appropriate agent.. for ex : s,1,Dial(ZAP/g4/5432) [g4 is the 4th span and 5432 is the ACD number of legacy pbx under which agents

Has anyone ever setup Asterisk to pass Feature Group D access while using a CIC code for outbound calls? If so can you please email the configuration you have done? I have tried to get this up and running but with no luck. I have also contacted support and I cant seem to get this going. Thanks in Advance, Jason Miller

Hello, I'm trying to run smbtorture against another system. I have installed version 4.0.0alpha9 locally. The remote system is registered with ADS as: distinguishedName: CN=bl-uits-cictest,CN=Computers,DC=ads,DC=iu,DC=edu name: bl-uits-cictest dNSHostName: bl-uits-cictest.ads.iu.edu servicePrincipalName: HOST/bl-uits-cictest.ads.iu.edu servicePrincipalName: HOST/BL-UITS-CICTEST The

I have a server with an asterisk ss7 link connected to a Siemens working well for over a year. A few days ago I started having problems with signaling. I found the following logs in / var / log / messages Sep 12 11:49:25 call3 kernel: [1018427.030959] dahdi: Master changed to TE2/0/2 Sep 12 11:49:25 call3 kernel: [1018427.120740] dahdi: Master changed to TE2/0/1 Sep 12 11:49:26 call3 kernel:

This patchset expands traditional CFQ scheduler in order to support cgroups, and improves old version. Improvements are as following. * Modularizing our new CFQ scheduler. The expanded CFQ scheduler is registered/unregistered as new I/O elevator scheduler called "cfq-cgroups". By this, the traditional CFQ scheduler, which does not handle cgroups, and our new CFQ

This patchset expands traditional CFQ scheduler in order to support cgroups, and improves old version. Improvements are as following. * Modularizing our new CFQ scheduler. The expanded CFQ scheduler is registered/unregistered as new I/O elevator scheduler called "cfq-cgroups". By this, the traditional CFQ scheduler, which does not handle cgroups, and our new CFQ

On Wed, Jun 04, 2014 at 01:18:47PM +0530, abhishek jain wrote: > Hi RIch > > Below are the necessary details of the system.. > > > *ls -l /boot/total 10200-rw-r--r-- 1 root root 5224765 Nov 22 2013 > uImage-rw-r--r-- 1 root root 5192861 Nov 22 2013 uImage.origls -l > /lib/modules/total 4drwxr-xr-x 3 ubuntu ubuntu 4096 Jun 3 07:21 > 3.8.13-rt9-QorIQ-SDK-V1.4uname

Hi, I'm planning to upgrade an old public/internal development infrastructure and will use CentOS 5.4 x86_64 as basis. One of the critical server is Subversion (as an Apache httpd module). We currently use Subversion v1.4 on the server and v1.6 on our clients. While I was very happy with the v1.4 server over the years, we now sometimes have weird issues which seem related to compatibility

Hi there, I'm experiencing an echo problem and dammed If I can sort it out. We're running Asterisk on Fedora Core 3 64bit, installed as per http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3. These are the specs of the Machine ? 1 x AMD A64/3500+ CPU: Desktop Athlon64? Retail w/fan SKT 1 x Asus A8N-SLI Deluxe Athlon? 64 S939 NVIDIA nForce(r)4 SLI? PCI Express Req: 24pin ATX 1 x

hello, all of users: sorry, resend it again for clarifying the message. I have implemented cha_ss7 in china. initially, the chan_ss7 can not support the call group. i modify the code. now the problem is that, both sides can hear the ring, but i can not hear the voice from each other. i think the ss7 does not send the voice steam to the destination. in chan_ss7, i added:

I am using an Asterisk box as a mini-softswitch and have run into a minor (hopefully) road block. The far end switch requires CIC (Carrier Identification Code) in the SIP invite like: INVITE sip:+18001234567;cic=+16789@hostname.com;user=phone SIP/2.0 ^^^^^^^^^^^^^^^^^^^^^^^ Is there a way to configure Asterisk to send this in the SIP invite? Any help would be

On Wed, Jun 04, 2014 at 03:29:11PM +0530, abhishek jain wrote: > supermin: kernel: SUPERMIN_KERNEL environment variable /boot/uImage > supermin: failed to find a suitable kernel (host_cpu=powerpc64). > > I looked for kernels in /boot and modules in /lib/modules. If you follow the code in src/kernel.ml you'll see that this means supermin was unable to get the kernel version from

On Wed, Jun 04, 2014 at 02:25:58PM +0530, abhishek jain wrote: > Hi RIch > > > Below are the details of the powerpc ubuntu system ... > > ls -l /boot/ > total 10200 > -rw-r--r-- 1 root root 5224765 Nov 22 2013 uImage > -rw-r--r-- 1 root root 5192861 Nov 22 2013 uImage.orig > > > ls -l /lib/modules/ > total 4 > drwxr-xr-x 3 ubuntu ubuntu 4096 Jun 3

Dear All I am trying to specify the output directory and the directory for the log files (the same) for texi2dvi. The default for my windows computer is C:\Users\...\Documents which I do not want. The help guide: texinputs NULL or a character vector of paths to add to the LaTeX and bibtex input search paths. is a little cryptic for me this morning I have tried texi2dvi(file =

Hi All, Eagerly waiting for v1.4.x to mature a bit before getting serious about it. Is it ready for production yet? If that's too general, where is it in terms of stability compared to where 1.2.x is now. Anyone running it successfully in production environment and if so what sort of config do you have?

I'm learning how to use tapply. Now I'm having a go at the following code in which dati contains almost 600 lines, Pot - numeric - are the capacities of power plants and SGruppo - text - the corresponding six technologies ("CCC", "CIC","TGC", "CSC","CPC", "TE"). .....................................................

I have four database servers in a high-availability, load-balancing configuration. Each machine has a mount to a common data source which is an OCFS2 v1.4 file-system. While working on three of the servers, I restarted the IP network and found after-wards the fourth machine hung. I could not reboot and could not unmount the ocfs2 partitions. I am pretty sure this was all caused by my taking down

I'm making outbound calls by placing call files in the asterisk outgoing directory. At times, the call would be hung by SS7 without even attempting (due to error in the outgoing number). I get the following on console: -- Attempting call on ss7/9297210213 for s at croom:1 (Retry 1) -- Sent IAM CIC=22 ANI=9134904821 DNI=9297210213 RNI= -- SS7 hangup 'SS7/callserver/22'

I have this [TRONCAL-SIP] exten=>225/91,1,Answer exten=>225/91,2,Echo exten=>225/91,3,Hangup exten=>225/92,1,Answer exten=>225/92,2,Playback(conf-invalid) exten=>225/92,3,Hangup When I make a call CLI> -- Recv IAM CIC=8 ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=>225,1,Answer Works ok -------------- next part --------------