search for: exvito

...ns -> ../init.d/xendomains - The behavior remains unchanged. This is looking much more of an upstart/sysvinit issue that a Xen 4.6 issue. However, I believe others may have been through this experience. Can anyone provide any pointers in further diagnosing this? Thanks in advance. Regards, -- exvito -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos-virt/attachments/20160407/2abcd774/attachment.html>

George, Thanks a lot for your investigation and feedback. I confirm it works as expected after the change you proposed. Cheers, -- exvito

...6.1-9.el6.x86_64 # rpm -V xen-runtime Confirmed installation of 4.6.1-9 from virt-xen-testing. Confirmed files "as packaged". # grep LOCK /etc/xen/scripts/hotplugpath.sh XEN_LOCK_DIR="/var/lock" The XEN_LOCK_DIR still points to the wrong directory. Am I missing something? -- exvito

...* kernel/wct4xxp/base.c: When doing the ISR rewrite, forgot to include the vpmdtmfcheck when doing DTMF polling causing it to check for DTMF events even when it was told not to (+others) I need to have this system running in about a week and a half. What do you guys say ? -- exvito

...here any way to setup the asterisk cli to use such keybindings ? I took a quick glance at 1.4.11 source and found readline.[ch] files, but asterisk is not behaving to my inputrc configuration... Googled for a while to no effect. Am I alone in this ? Thanks in advance for any hint, -- exvito

...entering music on hold (local, of course, after fixing my issues above!) and back to G.729 when back to conversation ? (ok, this probably needs patching the source !... but maybe someone has an idea or has taken a different approach at this...) :-) Thanks a lot for any feedback, -- exvito

On Thu, Apr 7, 2016 at 5:46 PM, exvito here <ex.vitorino at gmail.com> wrote: > Hello all, > > I'm addressing the mailing list following the suggestion of gwd on > #centos-virt @freenode IRC. > > The subject says most of it. Here are the details : > > - Was running 4.4.x and configured such that /etc/xe...

...just wait+hangup directly here!) System B runs asterisk 1.2.30.1 System B iax.conf: [one-systemA-user] type=user context=one-context notransfer=yes disallow=all allow=gsm System B extensions.conf: [one-context] exten => _NXXXXXXXX,1,Dial(.../${EXTEN}) exten => _NXXXXXXXX,n,Hangup() -- exvito

Hi All, I'm trying to achieve the following: - If <sip/iax user> logs in from home, they can dial internal extensions only (this is to avoid employees going wild on local/mobile calls from home) - If <sip/iax user> logs in from the office, they can call anyone they want. Since I have my users defined in an LDAP tree, I'd like to stick to one-account-per-user (each account is

On Mon, Apr 11, 2016 at 6:06 PM, exvito here <ex.vitorino at gmail.com> wrote: > George, > > Thanks a lot for your investigation and feedback. > I confirm it works as expected after the change you proposed. The patches have been backported and are available in 4.6.1-9 from virt-xen-testing. Please test it and report a...

On Tue, May 24, 2016 at 10:20 PM, exvito here <ex.vitorino at gmail.com> wrote: > On Tue, May 24, 2016 at 3:23 PM, George Dunlap <dunlapg at umich.edu> wrote: >> >> The patches have been backported and are available in 4.6.1-9 from >> virt-xen-testing. Please test it and report any problems here. >>...

hi list, I'm currently trying to get Asterisk running with an HFC-compatible ISDN card (no-name product, but supposed to work with Asterisk according to the packaging). the ISDN-card is connected to a alcatel ISDN-system where it should act just like a normal ISDN-phone. I went with http://www.misdn.org/index.php/MISDN_with_Asterisk and installed the package from beronet.com including:

Hello, I'm trying to set up call recording for a queue. Right now the recording appears to work correctly, but when I call and chatter for a minute or so, at the end of the call I end up with a very small file (less than 100 bytes), which contains about .06 seconds of silence. If I talk for another minute, this file will get up to 200 bytes or so. In my queue configuration, I have: [testq]

According to http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS when a caller hangs up before the callee has time to pick the phone up then DIALSTATUS should be CANCEL. And it is. However, the disposition field in the CDR table is "NO ANSWER". So if I analyze the CDR data I won't be able to discriminate calls cancelled by the caller and calls not answered by the callee

For high-availability Asterisk when using PRI Lines, you might also want to check out our product (FSV-4PFS). It's available at www.failsafevoip.com Bill Also to add to the last post does this device have hardware echo cancelation? if it does it could be a great replacement, if not may not be what I'd really want to use. Thanks, tom I've been looking at the RedFone foneBRIDGE2

Hi all, How do I set different rx and tx gains for each channel? in my zapata.conf file I have a heading [trunkgroups] and then [channels] under this I have information such as language context signalling etc and also rxgain and txgain. My assumption is that these settings are used for all channels (I have three zap channels). I need to have different gain settings on each channel. Is this

Running AsteriskNow, with FreePBX front end. Have two users who use softphones on notebooks in the field. Problem is that if the softphone receives a call, but the user is not available to pick it up, Asterisk will send the call to voice mail as normal, but the softphone will continue to ring...... I've tested the Virbiage and x-Lite softphones, same issue. Is there an IAX setting in

hi i am integrating asterisk whit an open source CRM or maybe i am integrating an open source crm whit asterisk any way the CRM is vtiger, (www.vtiger.com) i need an opensource queue log parser to put inside the crm agent statics. i need any parser i can port it to php or to java if you don't know any parser maybe you can send me a link or a pdf whit info on how to parse the log. Thanks David

Hi, I'd like to know what's the most "popular" method for automatic fax/voice detection for incoming calls on mISDN cards such as the B410P (hfcmulti). I'm running: kernel 2.6.17 misdn 1.1.3 asterisk 1.4.21.2 B410P card I'm using iaxmodem and hylafax with asterisk (the setup works for zap channels). I've used the following options in /etc/asterisk/misdn.conf:

Hello, Is it possible to retrieve the DIALSTATUS variable when placing call through a call file. This variable is set when using the Dial() application from the dialplan, but I am using a call file for my current application and need to get the dialstatus. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: