search for: asterisk11

...roblem 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 in which to save the key (/home/rsync11/.ssh/id_rsa): Created directory '/home/user1/.ssh'. Enter passphrase (empty for no passphrase): Enter same passphrase again: Your identification has been saved in /home/rsync...

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...

can i test my asterisk11 on a single machine on which asterisk is installed that sounds are working from both end properly. i have installed asterisk 11 on ubuntu12.04 with twinkle soft phone. regards abhi -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail...

>>>>> "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

...gt; 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 this set-up (Asterisk11 with > Snom870s using TLS) to work and if so could you provide the details? > > I have this in Asterisk sip.conf (loaded through FreePBXs > sip_general_additional.conf). > > tcpenable=yes > tlsenable=yes > tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt &gt...

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>

..."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 downloaded and built 11-beta1. Edited (according to the asterisk11 wiki-page) extensions.conf, chan_motif.conf, jingle.conf and restarted. Same behavior, except for minor details. As soon as I start, ejabberd tells me that the defined user becomes online. >From jitsi I can send...

...empting 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 this set-up (Asterisk11 with Snom870s using TLS) to work and if so could you provide the details? I have this in Asterisk sip.conf (loaded through FreePBXs sip_general_additional.conf). tcpenable=yes tlsenable=yes tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt tlscafile=/etc/pki/tls/certs/ca-bundle.cr...

A few months ago I started using and had to abandon PJSIP because my dialplan could not read the inbound signalling IP address, which I can read now in Asterisk11 using CHANNEL(recvip). My app relies on this information. The question is, is it possible now access the signalling IP of an incoming SIP call using PJSIP? Philip

Hi, all I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it "Asterisk11". I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database "mydatabase") via cdr_adaptive_odbc. The "SIP/A221" is anot...

Hi, all Sorry that forgot add mail subject last one. I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it "Asterisk11". I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database "mydatabase") via cdr_adaptive_odbc. The "SIP/A221" is anot...

Hi, all Sorry for null subject last mail. I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it "Asterisk11". I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database "mydatabase") via cdr_adaptive_odbc. The "SIP/A221" is anot...

Hi everyone! I tried to install chan_dongle for Asterisk 11 in a Ubuntu 14.04, but my Huawei E153 is not working in my Asterisk. I fallow this rules http://blog.denisbondar.com/post/asterisk11-chan_dongle_e1550-ubuntu14 But not successes. Thanks in advanced,

...mation from the SIP > signalling, or can you use AMI events for example? I think > this would be possible if asterisk is configured to stay in > the media path, so re-inviting is handled over asterisk itself > and therefore detectable with AMI events. > I am working with a FreePBX12/Asterisk11 setup. Asterisk stays on the path (B2B) and there are no peer-to-peer re-invites. What I am trying to do is to get our Snom870s to use a distinctive ring tone when external calls are transferred internally. I have an extension context override that detects the origin of calls and assigns a disti...

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 alterations to the default configuration, whatever it is. The asterisk host is...

exten => 179,1,SIPAddHeader(Call-Info:\;answer-after=0) exten => 179,2,Page(SIP/180&SIP/181&SIP/182&SIP/184) The asterisk11 page() application works great, but I've just learned that the person who initiated the page can transfer or conference the page if they don't hang it up before using those functions. It never would have occurred to me to try it, but a user did it accidentally today and it caused quite...

The sample config files in the Asterisk distribution and packages are really good for getting the demo up and running quickly, for example, to extend the demo to run behind a WebRTC proxy only required about 6 lines of extra code to define a peer in sip.conf and enable TCP However, I'm not sure that they should be installed by default by packages. Most package managers provide a way to diff

On Thu, March 5, 2015 05:30, Ruben R?gels wrote: > > > Am 05.03.2015 um 01:09 schrieb James B. Byrne: >> I am trying to determine how the transfer button on the Snom-870 >> works >> with Asterisk. Is the ## special code employed as when it is >> entered >> through the handset or is the blind transfer through the phone >> function accomplished in a