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Thanks for you answer Marcus, So maybe this means some bug was fixed? Anyone aware of something related? >From the release notes, I couldn't find any direct change that could fix this.... Thanks, Kind regards, Patrick Wakano On 21 February 2018 at 20:29, Marcus Kvarsell <Marcus.Kvarsell at fogwise.se> wrote: > Hello, i found upgrading to asterisk 15 helped. > > > > *Fr?n:* asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] *F?r *Patrick Wakano &gt...

...t_contents("https://blah.blah.blah", FALSE, @stream_context_create(array('https' => array('timeout' => 5, "verify_peer"=>false, "verify_peer_name"=>false)))); On 09/14/2018 01:40 PM, Carlos Chavez wrote: > On 9/13/2018 8:04 PM, Patrick Wakano wrote: > >> Hello list, >> Hope you all doing  well! >> >> Recently, I had an issue with a FastAGI PHP script, which under some >> specific situation would run into an infinity loop, consuming all CPU >> resources. This also was preventing Asterisk to termi...

Please can i ask you i want to know which code can help me to provide the taxation of voip/toip services in asterisk Le mar. 18 sept. 2018 à 01:36, Patrick Wakano <pwakano at gmail.com> a écrit : > Thanks everyone for the answers! > I did explored some options at the PHP level and probably will do > something in this direction, but in fact what I was really looking was > something in the Asterisk side, not in the script side. > Because...

...nds and so this would protect Asterisk preventing the dialplan to get stuck due to some external script problem that is actually outside of Asterisk control. Does Asterisk provide some control of this sort? I searched around and could not find any..... Any idea is appreciated! Kind regards Patrick Wakano -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180914/0bd1f571/attachment.html>

...e for a pure SIP environment the HANGUPCAUSE would have a more accurate information about the error. So question is can I always use this info and completely ignore what the DIALSTATUS is? Or does someone knows exactly where is more suitable to use one over the other? Thanks, Kind regards, Patrick Wakano -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180314/1b0c3a0b/attachment.html>

...he call itself... I've done some research and apparently the Monitor application would be able to record everything (but I didn't verify it myself).... Anyway why MixMonitor can't? Also does anyone have an idea on how to record everything in the same file? Thanks, Kind regards, Patrick Wakano -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180706/eb03c396/attachment.html>

...which causes this crash (happens around 2 times per month). Does anyone have any idea of what may be happening/wrong? I am running Asterisk version 13.6.0 on CentOS 6.6 kernel 2.6.32-504.el6.x86_64 and on CentOS 6.9 kernel 2.6.32-696.el6.x86_64. Any idea is very appreciated! Best regards, Patrick Wakano <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> <#DAB4FA...

...ion about when DIALSTATUS would actually be better. The HANGUPCAUSE was reworked in version 11 ( https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause) but I didn't find someone actually stating it is a better alternative or replacement to the DIALSTATUS or something similar. Cheers, Patrick Wakano On 14 March 2018 at 13:30, Dovid Bender <dovid at telecurve.com> wrote: > I would think that is a bug since the only time DIALSTATUS = BUSY is where > you got a 486 or 600 (as per https://wiki.asterisk.org/ > wiki/display/AST/Hangup+Cause+Mappings). > > On Tue, Mar 13, 2018...

...stutter 350+440/750,440/750 But the dialplan execution causes this: ERROR[20778][C-00000006]: func_channel.c:661 func_channel_write_real: Unknown country code '*gb*' for tonezone. Check indications.conf for available country codes. Any info is much appreciated! Cheers, Patrick Wakano <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> <#DAB4FAD...

Thanks very much for the reply Joshua! So I guess that setting dtmfmode=auto would be the safest choice in order to strip out the DTMFs from the recording, right? Cheers! Patrick Wakano On Tue, 1 May 2018, 19:36 Joshua Colp, <jcolp at digium.com> wrote: > On Mon, Apr 30, 2018, at 11:23 PM, Patrick Wakano wrote: > > Hello list, > > Hope you are all doing fine! > > > > I have stumbled over some piece of dialplan code in which apparently they > &...

...he tone as well, or is it only transmitted to the other leg without being generated to the recording file? Also, what if one or both legs are RFC2833? From my tests the RFC2833 events never show up in the recording, but I just want to confirm that this is always true. Thanks, Kind regards, Patrick Wakano <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> <#DAB4FA...

...g question from this, if the with-ssl is not mandatory to have the encryption support, what is it actually used for? Maybe it is some old flag which is not needed anymore and so can be ignored for now and possibly removed from the configure/makefile stuff for future releases? Kind regards, Patrick Wakano On Wed, 26 Feb 2020 at 06:33, Kevin Harwell <kharwell at digium.com> wrote: > On Thu, Feb 20, 2020 at 9:38 PM Patrick Wakano <pwakano at gmail.com> wrote: > >> Hello list, >> Hope you are all doing well! >> >> I am facing a problem when compiling Asterisk...

...-codec --enable-epoll checking for mandatory modules: PJPROJECT CRYPTO SRTP... ok Also, why am I having the pjsip startup errors when the --with-ssl is used? I could not find a clear explanation for this problem and how to fix it.... Any idea is much appreciated! Thank you, Kind regards, Patrick Wakano -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200221/5280ec9a/attachment.html>

...saying, so it is quite annoying but does not to cause issues: /usr/include/features.h:381:4: warning: #warning _FORTIFY_SOURCE requires compiling with optimization (-O) [-Wcpp]. Anyway, these problems do not happen if you manually build with the simple configure and make commands. Cheers, Patrick Wakano On Fri, 18 Oct 2019 at 11:54, Carlos Chavez <cursor at telecomab.mx> wrote: > They only problem I have found so far is while trying to install > Alembic for SQLAlchemy (for realtime configs). Those are the only packages > that I cannot get working properly. Vanilla Asterisk w...

...ory logs 3 times for this problem and they are similar so it seems that a bug exists somewhere leading to this problem. So anyone has any idea of what could be happening or if it may be related to some known bug? We are running Asterisk 13.6.0 in CentOS 6.6. Thanks very much! Best regards, Patrick Wakano -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20171114/84018d5c/attachment.html>

On Tue, Feb 25, 2020 at 4:02 PM Patrick Wakano <pwakano at gmail.com> wrote: > Hi Kevin! > Thanks very much for your reply! Much appreciated! > You're welcome! > So I just have a remaining question from this, if the with-ssl is not > mandatory to have the encryption support, what is it actually used for? > In As...

Hello Asterisk Users, Hope you all doing fine! I am working with a quite complex dialplan, and I've come to some situations where it makes some nasty use of pre-bridge handlers. The pre-bridge handlers wiki (https://wiki.asterisk.org/ wiki/display/AST/Pre-Bridge+Handlers) doesn't have the big warning the pre-dial one has indicating it must return and must not put the caller/callee in

On Thu, Feb 20, 2020 at 9:38 PM Patrick Wakano <pwakano at gmail.com> wrote: > Hello list, > Hope you are all doing well! > > I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and > I wonder if someone can put some light on it. > Log history short, install_prereq fails to install the packages (not s...

...ToS bits on UDP socket, ignoring TCP and TLS sockets (Reported by Joshua Roys) * ASTERISK-28812 - First DTMF is not get (Reported by Bernard Merindol) * ASTERISK-28758 - pjsip startup errors when using "with-ssl" configure option (Reported by Patrick Wakano) * ASTERISK-28824 - BuildSystem: Search for Python/C API when possibly needed only. (Reported by Alexander Traud) * ASTERISK-27717 - [patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7. (Reported by Alexander Traud) * ASTERISK-28798 - [patc...

...ToS bits on UDP socket, ignoring TCP and TLS sockets (Reported by Joshua Roys) * ASTERISK-28812 - First DTMF is not get (Reported by Bernard Merindol) * ASTERISK-28758 - pjsip startup errors when using "with-ssl" configure option (Reported by Patrick Wakano) * ASTERISK-28824 - BuildSystem: Search for Python/C API when possibly needed only. (Reported by Alexander Traud) * ASTERISK-27717 - [patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7. (Reported by Alexander Traud) * ASTERISK-28798 - [patc...