similar to: testing asterisk11 on single machine

I am experimenting with Asterisk having downloaded and installed the FreePBX i386 CentOS-6.3 based distro and updated it. The current package level on this system is: asterisk11-11.3.0-49_centos6 freepbx-2.11.0beta2-112 I am using twinkle-1.4.2-7.el6 as a softphone testing tool. There is no firewall on the asterisk host and SELinux is disabled on it. Fail2Ban is installed but I have made no

Am 03.03.2015 um 18:16 schrieb James B. Byrne: > CentOS-6.5 (FreePBX-2.6) > Asterisk-11.14.2 (FreePBX) > snom870-SIP 8.7.3.25.5 > > I am having a very difficult time attempting to get TLS and SRTP > working with Asterisk and anything else. At the moment I am trying to > get TLS functioning with our Snom870 desk-sets. And I am not having > much luck. > > Since this

Dear List wanna configure click2call in such a manner that my asterisk box call two mobile numbers and connect both numbers to talk. I have configured voip gateway, my internal and external calls are working fine. please help , abhi -------------- next part -------------- An HTML attachment was scrubbed... URL:

*Dear List* Plz help, i am not much experienced with asterisk. i configured it on ubuntu 12.04. no problem when i call any mobile no(0091XXXXXXXXXX) but when i call on my local asterisk no.(101,102 or 105) it is not connecting giving error "Auto fallthrough, channel 'SIP/lucknow-0000006f' status is 'CHANUNAVAIL' *while when i call 200 it is playing audiofile successfully.

CentOS-6.5 (FreePBX-2.6) Asterisk-11.14.2 (FreePBX) snom870-SIP 8.7.3.25.5 I am having a very difficult time attempting to get TLS and SRTP working with Asterisk and anything else. At the moment I am trying to get TLS functioning with our Snom870 desk-sets. And I am not having much luck. Since this is an extraordinarily (to me) Byzantine environemnt I am going to ask if any of you have gotten

Hi!I've a problem using ssh without password: I want use rsync for automatic scripts,I'm using this 2 names for my asterisk@home2.5 linux (based on red hat), rsync11 and rsync12. This is the way I use to change the configuration and then using without password , but the password is always asked: [rsync11@asterisk11]$ ssh-keygen -t rsa Generating public/private rsa key pair. Enter file

Hi! I' ve this problem: I've 2 asterisks box, asterisk11 and asterisk12, and one wi_fi phone. I call from wi_fi to a X-lite phone on a windows xp.I've setuped the X-lite to my vrrp IP (vrid IP) and the call is ok, I call from the wi_fi to X-lite and from the X-lite to wi_fi. In asterisk panell is all ok, and I listen the voice to the xp and in the wi_fi phone. asterisk12 is my master.

>>>>> "JBB" == James B Byrne <byrnejb at harte-lyne.ca> writes: JBB> tcpenable=yes JBB> tlsenable=yes JBB> tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt JBB> tlscafile=/etc/pki/tls/certs/ca-bundle.crt JBB> tlsdontverifyserver=yes JBB> tlscipher=ALL JBB> tlsclientmethod=tlsv1 You are missing the tls key. The config name is

Other things to consider: The transport config, which can be in [general] or in a peer's [] block. if you want tls-only, use transport=tls it also accepts tcp, udp or a comma-separated list. if given a list, it tries them in order If you need ast to register over tls, use something like this: register => tls://username:xxxxxx at sip-tls-proxy.example.org (copied from the

This seems to me to be getting down to some sort of problem with configuring the Snom-870. when I register the device 41712 (set up for transport=tls only) then I see this in the SIP trace: Sent to udp:192.168.6.9:5060 at 4/3/2015 09:07:36:813 (836 bytes): REGISTER sip:voinet09.internal.hamilton.harte-lyne.ca:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.6.112:5060;branch=z9hG4bK-udx92poqese6;rport

I tried debugging it and the issue seems to be in the implementation of MachineInstrIterator.h and the way it interacts with GraphWriter.h functions. I found this by replacing the ( template <> struct GraphTraits<MCDFGraph<MachineFunction*> >) with a similar MCDFGraph based template of CFG similar to the one in MachineFunction.h (similarly replacing the DOTGraphTraits with the

On Tue, March 3, 2015 13:37, James Cloos wrote: >>>>>> "JBB" == James B Byrne <byrnejb at harte-lyne.ca> writes: > > JBB> tcpenable=yes > JBB> tlsenable=yes > JBB> tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt > JBB> tlscafile=/etc/pki/tls/certs/ca-bundle.crt > JBB> tlsdontverifyserver=yes > JBB>

Hi, I am trying to generate the DFG of machine functions. Initially, I added a pass to generate the DFG of LLVM IR functions. This was based on the mail thread - http://lists.cs.uiuc.edu/pipermail/llvmdev/2009-September/025582.html. This pass worked fine and I was able to generate DFG of LLVM IR functions. Later, I ported the DFG pass code for machine functions. I ported the InstIterator.h

Hi, I'm using qemu+ssh://username at hostname/system as the remote URI. Libvirt seems to be communicating fine until some 2 minutes (we poll every 5 seconds) and then it throws up RPC error and many counters are wrong. But if I collect on localhost the counters seems to be coming fine. Test and testvm are guests on the local machine where as Win8 and Ubuntu* are remote URI's connected

Happy new 2014, just saw on <heise.de> the announce of LLVM v3.4. The sig-file for the pre-built binary for Ubuntu/precise seems to point to the tarball. Can someone fix that, please. Thanks. Regards, - Sedat - P.S.: Some wget outputs. $ LC_ALL=C wget http://llvm.org/releases/3.4/clang+llvm-3.4-x86_64-unknown-ubuntu12.04.tar.xz.sig --2014-01-07 10:39:22--

On 11/17/2013 09:18 PM, Matthew Harrold wrote: > On Sun, Nov 17, 2013 at 7:44 PM, Cole Robinson <crobinso@redhat.com > <mailto:crobinso@redhat.com>> wrote: > On 11/17/2013 01:26 PM, Matthew Harrold wrote: > > Hi All. > > > > I'm trying to setup a guest within my hypervisor that acts as a Console > > Server, using 10 USB

Hello, I have an error while try to install twinkle: # yum install twinkle [...] Resolving Dependencies --> Running transaction check ---> Package twinkle.i386 0:1.2-1.el5.rf set to be updated --> Processing Dependency: libresolv.so.2(GLIBC_PRIVATE) for package: twinkle --> Finished Dependency Resolution Error: Missing Dependency: libresolv.so.2(GLIBC_PRIVATE) is needed by package

-- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-088e6ea8", "conversation to GSM") in new stack -- Executing [0473775006 at intern:2] Dial("SIP/twinkle-088e6ea8", "SIP/3starsnet/0473775006") in new stack -- Called 3starsnet/0473775006 -- Got SIP response 482 "Loop Detected" back from 85.119.188.3 -- Now forwarding

Hi all, For one of my inverstigations it looks like i'm back to "square one" I'm trying to accept an incoming xmpp call and forward it conditionally to a sip, isdn, or voicemail. No google is involved as i use a local xmpp server (ejabberd) I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but some suggested me to have a look at asterisk11,so i did... I

Hi All. I'm trying to setup a guest within my hypervisor that acts as a Console Server, using 10 USB to Serial adapters. The Host is all setup and can access each Serial Port without problems. I have attempted to add the Serial Connections to a Guest, using both Virtual Machine Manager and by editing the .XML file, and I can add up to 4 Serial devices (Including one for console access to the