similar to: Is svn.asterisk.org down ?

Under Firefox, browsing to https://reviewboard.asterisk.org I get Warning: Potential Security Risk Ahead Firefox detected a potential security threat and did not continue to reviewboard.asterisk.org. If you visit this site, attackers could try to steal information like your passwords, emails, or credit card details. Websites prove their identity via certificates, which are issued by certificate

Hi Nick, I was not aware of it, but it makes sense given the recent switch to ORC2, which has different symbol resolution rules. I am out on vacation this week, but will take a look when I get back and see if I can restore the old behavior. Cheers, Lang. Sent from my iPhone > On Jan 20, 2019, at 2:14 PM, David Blaikie <dblaikie at gmail.com> wrote: > > +Lang who does JIT

Hello. I would like to run asterisk as an user other than root. I have seen some tutorials on the web, but I would like to know if there is some ?official? how-to for this. Is there? I looked at a thread on reviewboard regarding this (https://reviewboard.asterisk.org/r/654/). It was Paul Belangers work trying to make the installation process take care of this. But the conclusion seem to

I'm trying to install Review Board (http://review-board.org) with python-setup tools and "sudo easy_install ReviewBoard", but this fails as follows: easy_install ReviewBoard Searching for ReviewBoard Best match: ReviewBoard 0.9.dev-20081202 Processing ReviewBoard-0.9.dev_20081202-py2.4.egg ReviewBoard 0.9.dev-20081202 is already the active version in easy- install.pth

Not yet unfortunately. I've had my head down working on a jut-linker replacement. Let me take a look right now... On Mon, Feb 18, 2019 at 10:40 AM David Blaikie <dblaikie at gmail.com> wrote: > Ping - did this end up getting addressed? > > On Mon, Jan 21, 2019, 6:15 PM Lang Hames <lhames at gmail.com wrote: > >> Hi Nick, >> >> I was not aware of it,

I have expanded the EWS calendar functionality within Asterisk 1.8 so it is now possible to access any calendar within an Exchange 2007 or 2010 server. I have put the changes onto the reviewboard for astrisk but currently no one responded. So if you use the EWS calendar functionality within Asterisk and would like to have access to any calendar in Exchange please try the patch in the

Hello List, I have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Before continuing, this is my environment: Asterisk: 1.6.2.16.1 OS: CentOS release 5.5 (Final) 2.6.18-194.32.1.el5 Details: I have this block on sip.conf ----- start ---- ... register => john:j0nhp4ss

Good morning, There is a new patch up on reviewboard[1] right now for the removal of app_rpt and chan_usbradio from Asterisk trunk. As it stands right now these two modules do not appear to be maintained in this repository and have out-of-date code. Russellb's patch will see these to modules removed from asterisk trunk (asterisk 11). If a large part of the community wishes to help

Hi, I didn't notice on my first answer, but we are on the -dev list and this is not related to asterisk code developing. I will answer you on the -users list, so we can continue the discussion there. Cheers, -- Ing. Miguel Molina Grupo de Tecnolog?a Millenium Phone Center Anahi Ludue?a escribi?: > Hi, thanks Miguel. > I have another question: if I want to call the GoSub

Hi all, I'm running old Sernet samba 4.0.9 on Debian and trying to set up LDAP authentication for https://www.reviewboard.org/docs/manual/3.0/admin/configuration/authentication-settings/ To cut a long story short about half of users can log in and half not without any obvious reasons that ldapsearch comparisons would reveal. So I really want to see what the server is saying. I've

>>> Reviewboard is a legacy site and will likely be shutdown. Is there a reason you are wanting to visit it? After seeing Olivier's post about his recent failures on compile and it referencing NBS (Network Broadcast Sound), which I had never heard of, I was googling to find out more and that was one of the Google hits Doug

Hi Richard > That could be possible and would be a bug in chan_sip. Ok, so I switched to PJSIP to see if this behaves differently So ip do a Transfer(PJSIP/${DESTNUMBER}@trunk) And this results in: Failed to parse destination URI '[destnumber scrubber]' for channel PJSIP/trunk-00000011 Do I have to specify the destination number differently when using Transfer with pjsip that I

hi, is there somebody using systemd start script on fedora/centos7 + asterisk 13 in production? i have strange problem with high cpu usage when asterisk is started via systemd thanks for feedback p.s. systemd script is not in vanilla asterisk. only in fedora package info https://reviewboard.asterisk.org/r/2730/ -- --------------------------------------- Marek Cervenka

We have a server that records all calls so we set Mixmonitor with the b option to only record calls that are actually bridged. I notice that we have lost of 44 byte files in /var/spool/asterisk/monitor which correspond to calls that were not answered. If a call is not answered I assume it was never bridged so why would Asterisk create a file? Is there a way to avoid getting those empty

Hi everyone, I was just wondering, does anyone managing production asterisk servers migrated successfully from 1.4.2X to 1.6.0.X? I would like to see if there are some successful cases. Is your 1.6.0.X behaving well, with acceptable stability? Please share your experiences. Thanks, -- Ing. Miguel Molina Grupo de Tecnolog?a Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57

Hi , I am trying to install office XP under wine build 20020904 .Before anything could happen the setup.exe tells me that I have to run win98 ,win millenium ,win 2000 or latter to use office xp . Have I got to change anything in my config file or perhaps should I use wineX ? Thanks to All . Julien -- Julien Motch <julienmotch@skynet.be> -------------- next part -------------- A

Hi all, I'm testing the persistent dynamic queue members functionality on 1.6.0.10. The queue members are agents defined in the agents.conf file. When I issue an asterisk restart and check the queue members again on the CLI, all of them are listed as /invalid/ and there is no way to change this but to unload app_queue.so and load it again. My guess is that the internal AstDB queue

Hi Guys I am new working with lumenvox products, and unfortunately I had not been able to install it properly, I follow the steps in lumenvox site and it looks like it works I mean: ========================================================= [root at pbx-millenium examples]# ./example 127.0.0.1 Connecting to 127.0.0.1 Interpretation 1: 8587070707 count=0, decode returns 1 Interpretation 1:

We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13 The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is supported, but when I enable it I get the message "res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored." Is v34 only supported with SpanDSP? Also, the res_fax.conf.sample does not indicate v34 as a valid

I'm trying to use Asterisk as a PSTN simulator to run performance tests for echo cancellation algorithms. I'm using the following configuration: SIP <-----> Asterisk 1 <----> Asterisk 2 <----> Echo() Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan application. The problem is the high delay using this configuration: 20 ms only in Asterisk 2.