search for: pwakano

...vid 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 at 10:11 PM, Patrick Wakano <pwakano at gmail.com> > wrote: > >> Hello list, >> Hope all doing well! >> >> I've been checking some cases when a Dial fails and dialplan execution >> continues to handle this. I am finding it a little confusing how we should >> handle the DIALSTATUS and th...

...h 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 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...

Hello list, Hope all doing well! I've been checking some cases when a Dial fails and dialplan execution continues to handle this. I am finding it a little confusing how we should handle the DIALSTATUS and the HANGUPCAUSE in this situation.... More specifically, I am facing a case in version 13.6.0 where I am getting a DIALSTATUS=BUSY and HANGUPCAUSE=19 after receiving a 480 SIP error. Seems

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 in my opinio...

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 Asterisk is al...

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

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 sure how important they actually are....): speexdsp-devel, gmime-devel, uriparser-devel, iksemel-devel, uw-imap-devel, hoard Then, I am running the following commands

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 sure > how...

Hello list, Hope you all doing fine! I've tried to use the 'alias' directive in the indications.conf file but apparently it doesn't work.... It looks like maybe this feature was removed, because old sample for the indications.conf file have example using the alias parameter, but newer samples don't have it anymore.... also I couldn't find any ticket saying this parameter

Hello Asterisk list, I've facing a memory allocation issue that happens occasionally but on a consistent basis. The problem happens as follow, suddenly Asterisk starts consuming a lot of memory, in a rate of more than 1GB per hour. Kernel will eventually kill it via the OOM killer when memory is really exausted... This situation does not generate backtrace because Asterisk is responsive

Hello Asterisk list, Hope you are all doing well! We are using the MixMonitor application to record the calls and under some situations the call can be spied using ChanSpy with whisper enabled. Sometimes the spying channel is a person who can interact in the call, and some other times it is a sound file playing a message. The problem is that for some reason the MixMonitor does not record whatever

I don't know AGIspeedy, but I have some PHP scripts where I set a connect timeout using streams. Example using https, but should be easily adaptable to non-s http.: $pbxsh_bin = @file_get_contents("https://blah.blah.blah", FALSE, @stream_context_create(array('https' => array('timeout' => 5, "verify_peer"=>false,

Hello George, Hope this finds you well! I wonder if there has been any progress on this matter? I've been working to have Asterisk running on CentOS 8 and our jump from CentOS 6 to 8 doesn't look too bad.... The missing packages found are: gmime-devel, iksemel-devel, corosynclib-devel, libresample-devel, hoard and python-devel. Python-devel could be replaced by python2-devel or

Hello, I'm familiar with Pickup/PickupChan for taking a ringing call, but does anyone know how a phone can "steal" an already answered call from another phone? Our users have decided that call parking is too long-winded and don't want to use that. For example: phone A calls phone B, phone B answers the call, phone C dials something to "steal" the call from B, and

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! > > >

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 terminated the call properly because it was waiting for the AGI to return... The application uses AGIspeedy to process the AGI calls, not sure if this can be affecting this

Hello list, Hope you are all doing fine! I have stumbled over some piece of dialplan code in which apparently they were trying to avoid recording the DTMF tones in the wav file. It is really messy and I am not sure if this really works. So after a bit of research I found this comment ( https://community.asterisk.org/t/asterisk-dtmf-record/65040) in which it is said: *"Asterisk strips the

Hello Asterisk list, I am facing some Asterisk crashes which are consistently pointing to the same backtrace, which is the following (using DONT_OPTIMIZE, BETTER_BACKTRACES and MALLOC_DEBUG): Thread 1 (Thread 0x7f1f08be8700 (LWP 1767)): #0 0x00007f1f9bed3395 in __strcasecmp_l_sse42 () from /lib64/libc.so.6 #1 0x00000000004a91ca in cdr_object_get_by_name_cb () #2 0x0000000000463c60 in

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. > > > >