Re: [asterisk-users] Patch to remove numbers from the logs

If you need something quick you could create a batch script with sed or awk to remove the log lines you want and attach it to the prerotate script of logrotate (in case you use any of these in your env). Certainly this is not a final solution but it is already something that doesn't depend on an

Re: [asterisk-users] Asterisk and CentOS 8

In case anyone out there is working with CentOS, you might reconsider that decision: https://blog.centos.org/2020/12/future-is-centos-stream/ On Mon, 11 May 2020 at 23:53, George Joseph wrote: > > > On Sun, May 3, 2020 at 6:07 PM Patrick Wakano wrote: > >> Hello George, >

Re: [asterisk-users] Exceptionally long queue length queuing

inphone just press the hold once and then press and hold the spacebar which will repeatedly do the hold and unhold several times). After Wait is finished the "Exceptionally long queue length queuing" will show up but gets resolved very fast just because I think there weren't enough frames qu

Re: [asterisk-users] Asterisk and CentOS 8

ying 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

Re: [asterisk-users] pjsip startup errors when using "with-ssl" configure option

That makes sense Kevin! Thanks for the explanation, I will create a ticket for this then! Kinds regards, Patrick Wakano On Wed, 26 Feb 2020 at 09:33, Kevin Harwell wrote: > On Tue, Feb 25, 2020 at 4:02 PM Patrick Wakano wrote: > >> Hi Kevin! >> Thanks very much fo

Re: [asterisk-users] pjsip startup errors when using "with-ssl" configure option

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 wrote: > On Thu, Feb 20, 2020 at 9:38 PM Patrick Wakano wrote: > >> Hello list, >> Hope you are all doing well! >> >> I am f

[asterisk-users] pjsip startup errors when using "with-ssl" configure option

sl 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Audit AMI actions commands

connected via AMI and I want it logging all the actions (and the details of these actions) it was asked to do. Any idea is much appreciated! Thanks, Kind regards, Patrick Wakano -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] How best to run a SIPp test on a remote host

I've never worked with fabric nor pysipp, but if you want to run sipp in background so you can log out from your remote server and leave the test running, I suggest you can use screen. It works well for me. Patrick Wakano On Sat, 20 Oct. 2018, 01:55 Olivier, wrote: > Hello, > > I'm

Re: [asterisk-users] AGI timeout option

itself should be able to timeout a long running script and return to the dialplan. However looks like there is nothing of this sort. Kind regards, Patrick Wakano On Sat, 15 Sep 2018 at 03:56, Eric Wieling wrote: > I don't know AGIspeedy, but I have some PHP scripts where I set a > c

[asterisk-users] AGI timeout option

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

[asterisk-users] 400 reply to INVITE not properly treated

behaviour? I could not find anything related In my opinion, Asterisk should at fail the Dial and proceed with whatever was configured in the dialplan I tried some other 4XX SIP codes, but the only one I found not behaving properly is the 400 one Thanks, Kind regards, Patrick Wakano

Re: [asterisk-users] How to steal an answered call?

By the way, bear in mind this is exactly what a blind transfer from B to C would do, but with a lot of more work On Mon, 9 Jul. 2018, 16:36 Patrick Wakano, wrote: > Not sure how elegant this is, but I think you can try to elaborate some > logic that when phone C dials something, it

Re: [asterisk-users] How to steal an answered call?

Not sure how elegant this is, but I think you can try to elaborate some logic that when phone C dials something, it would retrieve you the channel phone A is connected and use the Bridge application to force the connection of phone C to phone A. So you need first to save the channels you have

[asterisk-users] MixMonitor multiple times to the same file

opened by some other MixMonitor thread. Or is there any reason/situation in which this is not desired? Kind regards, Cheers, Patrick Wakano -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check

Re: [asterisk-users] MixMonitor and ChanSpy whisper

Thanks for the reply Joshua! I did look the code, but it's too complicated for my old C knowledge :( So I guess I am left with the Monitor app Also a ConfBridge would also work instead of ChanSpy with whisper Cheers, Patrick Wakano On 6 July 2018 at 08:51, Joshua Colp wrote: > On

[asterisk-users] MixMonitor and ChanSpy whisper

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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] MixMonitor recording when in the holding bridge

ld be done? If not automatically maybe with some new flag Thank you, Kind regards, Patrick Wakano -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk

Re: [asterisk-users] DTMF tones in MixMonitor recording

I agree! I have my SBC and asterisk servers all configured with rfc2833, so it should be ok! No need for auto mode! Thanks again! Cheers Patrick On Tue, 1 May 2018, 20:07 Joshua Colp, <jc...@digium.com> wrote: > On Tue, May 1, 2018, at 6:52 AM, Patrick Wakano wrote: > > T

Re: [asterisk-users] DTMF tones in MixMonitor recording

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, <jc...@digium.com> wrote: > On Mon, Apr 30, 2018, at 11:23 PM

[asterisk-users] DTMF tones in MixMonitor recording

he 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_source=link_campaign=sig-email_content=webmail> Virus-free. www.avg.com <http://www.avg.com/email-s

Re: [asterisk-users] Alias for country in indications.conf

Just did Tzafrir suggestion and it worked like a charm! Thanks very much! Cheers, Patrick Wakano <http://www.avg.com/email-signature?utm_medium=email_source=link_campaign=sig-email_content=webmail> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email_source=link

Re: [asterisk-users] Alias for country in indications.conf

That's quite interesting Tzafrir! I will give it a try! Thank you very much for the idea! Cheers, Patrick Wakano On 23 April 2018 at 23:42, Tzafrir Cohen <tzafrir.co...@xorcom.com> wrote: > Also, > > On Mon, Apr 23, 2018 at 04:08:58PM +1000, Patrick Wakano wrote: > > Hel

Re: [asterisk-users] Alias for country in indications.conf

Hello Richard! Thanks very much for your answer! It all makes sense. I will just duplicated the tones config then! Thanks again! Cheers! Patrick Wakano On 23 April 2018 at 23:17, Richard Mudgett <rmudg...@digium.com> wrote: > It looks like any support for "alias" as a tone

[asterisk-users] Alias for country in indications.conf

lplan execution causes this: ERROR[20778][C-0006]: 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=e

Re: [asterisk-users] DIALSTATUS vs HANGUPCAUSE

That's really good info Tony! Thanks very much for the response! I will consider this to implement a better approach for the failed cases! Cheers, Patrick Wakano On 14 March 2018 at 20:44, Tony Mountifield <t...@softins.co.uk> wrote: > In article <CAPu3kNV8w+bYQT0W+QbnTSby0V5gfjLqZ

Re: [asterisk-users] DIALSTATUS vs HANGUPCAUSE

://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 <do...@telecurve.com> wrote: > I would think that is a bug since

[asterisk-users] DIALSTATUS vs HANGUPCAUSE

knows exactly where is more suitable to use one over the other? Thanks, Kind regards, Patrick Wakano -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https

Re: [asterisk-users] Asterisk crash on core show channel

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

[asterisk-users] Asterisk crash on core show channel

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/em

Re: [asterisk-users] RTCP + Stasis causing high memory consumption

Apparently it is an internal only problem, not actually outputting abnormal traffic in the network Best regards! Patrick Wakano On 16 November 2017 at 02:34, Matthew Jordan <mjor...@digium.com> wrote: > > > On Mon, Nov 13, 2017 at 11:42 PM, Patrick Wakano <pwak...@gmail.com>

[asterisk-users] RTCP + Stasis causing high memory consumption

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

Re: [asterisk-users] Surrogate channels

Thanks very much for the explanation Richard!! Best Regards! Patrick Virus-free. www.avg.com

[asterisk-users] Surrogate channels

Hello Asterisk list! I've been facing some scenarios in my dialplan where I see the "h" extension being executed for Surrogate channels. For me, it is kind of a mystery what these Surrogate channels are... I couldn't find good information about them... the source code is where I could find the

[asterisk-users] Asterisk crash in ast_find_ourip

Hello list, We've got an Asterisk crash in one of our servers and the core dump showed following call tree. Is this anyhow helpful to someone? Seems like a regular RTP / RTCP handling that lead to a malloc crash Grateful for any help! Cheers, Patrick Thread 1 (Thread 0x7f8d6b023700 (LWP

Re: [asterisk-users] Execution of pre-bridge handlers

What an excellent response Richard!!! Thank you very much for that!! Best regards! Patrick On Wed, Feb 15, 2017 at 5:23 AM, Richard Mudgett <rmudg...@digium.com> wrote: > > > On Tue, Feb 14, 2017 at 6:24 AM, Patrick Wakano <pwak...@gmail.com> wrote: > >> Hello Aste

[asterisk-users] Execution of pre-bridge handlers

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