similar to: rotatestrategy = none not working

Hello, We have an Asterisk 11.3 server where we want log rotation handled purely by Linux's logrotate, and not by Asterisk. To this end we've configured the [general] action of /etc/asterisk/logger.conf with: rotatestrategy = none However, an "asterisk -rx 'logger reload'" still rotates the log files. Does anyone know why? Thank you in advance, -- David Cunningham,

On Wed, 20 May 2020, David Cunningham wrote: > Thanks for the answer. Since that's what we already have configured, any > idea why it wouldn't work? As I said, when "asterisk -rx 'logger > reload'" is run it still rotates the log file. Sorry. No clues. I always use 'syslog' for logging everything. I just did a quickie test to see if I could replicate

Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording.

Hello, We have a need to record audio and allow the user to press any DTMF key to end the recording. Currently we're using the AGI command "record file" which does allow us to specify which DTMF keys can end the recording. However we also need to know *which* key actually ended the recording. Note that only allowing # or * to end the recording won't work for us. Does anyone

Hi Dovid, We can change the SDP in Kamailio, but Asterisk will still send its RTP from its default address. The remote end is strict about accepting RTP from the specified source and won't accept it. Have you any suggestions to solve that problem? Thank you. On Fri, 30 Oct 2020 at 14:49, Dovid Bender <dovid at telecurve.com> wrote: > Why not use OpenSips/Kamailoo in between?

Hello, Does anyone know a way with chan_sip to tell Asterisk to use a specific IP address for its end of the communication for a specific device? Something like: [device] type = friend host = 11.22.11.22 ouraddress = 33.44.33.44 This is for use on a server with multiple IP addresses. There is the "extenip" setting, but it's really designed for NAT, and can only appear in the

Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio On Thu, Oct 29, 2020 at 20:44 David Cunningham <dcunningham at voisonics.com> wrote: > Hello, > > Does anyone know a way with chan_sip to tell Asterisk to use a specific IP > address for its end of the communication for a specific

Hi David, It seems like a network issue to me, As it's unable to connect the other node and getting timeout. Few things you can check- * Check the /etc/hosts file on both the servers and make sure it has the correct IP of the other node. * Are you binding gluster on any specific IP, which is changed after your update. * Check if you can access port 24007 from the other host. If

OK, thank you George. On Sat, 24 Oct 2020 at 03:16, George Joseph <gjoseph at digium.com> wrote: > > > On Thu, Oct 22, 2020 at 4:13 PM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hi George, >> >> Thank you for the response. I'm a little unclear on what you mean by a >> transport. We're using chan_sip, not pjsip.

I don't know in that specific output what happened. Your best course of action is to add further logging or step through the logic with all of the knowledge you have of the RTP streams to understand what is happening. On Mon, Apr 17, 2023 at 8:52 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Thank you for that. From the code it kind of looks like

Hi Joshua, Thank you for that. From the code it kind of looks like STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum: if (!ast_sockaddr_isnull(&rtp->strict_rtp_address) && STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(), rtp->rtp_source_learn.start)) { ast_verb(4, "%p -- Strict RTP learning complete - Locking on source address %s\n", Our call shows: #

On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi George, > > Thank you for the response. I'm a little unclear on what you mean by a > transport. We're using chan_sip, not pjsip. > > Do you mean a device in sip.conf, using bindaddr to set the address to > bind for that device? We've only used bindaddr in the

Hi George, Thank you for the response. I'm a little unclear on what you mean by a transport. We're using chan_sip, not pjsip. Do you mean a device in sip.conf, using bindaddr to set the address to bind for that device? We've only used bindaddr in the [general] section before, but if it will work in a device that could be the answer. On Fri, 23 Oct 2020 at 00:13, George Joseph

It's probably best if you read the logic[1]. There's an entire comment that talks about how it works. [1] https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158 On Mon, Apr 17, 2023 at 7:10 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Could you confirm if the 5 second period for learning a new audio stream > is a minimum

Hi Joshua, Could you confirm if the 5 second period for learning a new audio stream is a minimum or a maximum? The unusual call flow in question results in Asterisk learning a new audio stream when we don't want it to, and having a minimum of say 2 seconds of audio would help avoid this. Thank you! On Thu, 2 Mar 2023 at 12:32, Joshua C. Colp <jcolp at sangoma.com> wrote: > On

Hi David, On Dec 24, 2019 02:47, David Cunningham <dcunningham at voisonics.com> wrote: > > Hello, > > In testing we found that actually the GFS client having access to all 3 nodes made no difference to performance. Perhaps that's because the 3rd node that wasn't accessible from the client before was the arbiter node? It makes sense, as no data is being generated towards

Hi David, Gluster supports live rolling upgrade, so there is no need to redeploy at all - but the migration notes should be checked as some features must be disabled first. Also, the gluster client should remount in order to bump the gluster op-version. What kind of workload do you have ? I'm asking as there are predefined (and recommended) settings located at /var/lib/gluster/groups . You

Hello all, Voisonics is pleased to introduce easySysAdmin, an automated support/security platform, designed to save your engineer's time and prevent hacking attempts and telecom fraud. It comprises of an online service run by us, and a lightweight and easy-to-install client on your side. Specifically of interest to Asterisk users is the monitoring of SIP registrations, and automatic blocking

Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the "enable video" checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with "Rejecting secure video stream without encryption details". - sipML5, but it won't register, perhaps something to do with not using the Asterisk

Hello, We have a system that interoperates with an external service, so that the basic call flow is: PSTN origination -> Asterisk A -> External service -> Asterisk B Initially the SDP from the external service tells the two Asterisks to send RTP directly to each other. Part way through the call the external service sends re-INVITEs both Asterisks to change the address for audio to