similar to: Asterisk 13 and Asterisk 17 Security Fix Only

Thanks Joshua for replying ! What would you advise : - leaving STUN address empty, in rtp.conf, as "STUN is not required for ICE" - configure it with one public STUN (I'm using stun.voip.ovh.net for this but I don't know how this server really works) Cheers Le mar. 27 oct. 2020 à 09:53, Joshua C. Colp <jcolp at sangoma.com> a écrit : > On Tue, Oct 27, 2020 at 5:35 AM

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

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, 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: #

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

On Tue, Jul 11, 2023 at 3:40 PM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, Jul 11, 2023 at 3:38 PM TTT <lists at telium.io> wrote: > >> That answers part two…but is there any mapping of AMI version to Asterisk >> versions? >> > > No, there is not. > I can say that Asterisk 13 is 2.x.x though because I just looked, so you can use the

On Tue, Feb 28, 2023 at 9:51 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, Feb 28, 2023 at 9:50 AM David Cunningham < > dcunningham at voisonics.com> wrote: > >> Hello, >> >> Does anyone know if one of the "strictrtp" options disables RTP learning? >> As far as I can tell from the documentation the values "no" and

Josh, What should Asterisk do if one of the real time methods fail? I have in extconfig.conf musiconhold => curl,http://localhost/moh.php,1 musiconhold => mysql,db-east,asterisk_moh,2 If the first server sends back a 404 it does not go to the second connection. Shouldn't a 404 be considered a failure and it should then move over to the next rt engine? On Thu, Jan 2, 2020 at 7:06 AM

Is it possible to disable the unbond resolver in the asterisk configuration? Or, it is necessary just to disable the module? Best regards, Leonid Fainshtein On Thu, Aug 27, 2020 at 3:29 PM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Thu, Aug 27, 2020 at 9:24 AM Leonid Fainshtein < > leonid.fainshtein at xorcom.com> wrote: > >> I deleted the

Thanks Joshua, I assume by query asterisk you mean I'll need to query it via AMI? Is that information available via AMI? *Nick Olsen* Network Engineer Office: 321-408-5000 x103 Mobile: 321-794-0763 On Tue, May 26, 2020 at 2:57 PM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, May 26, 2020 at 10:48 AM Nick Olsen < > nick at floridavirtualsolutions.com> wrote: >

I deleted the res_resolver_unbound.so module, and now it works as expected. So, the problem is related to the 'unbound' resolver? FYI: I'm using Asterisk 16.2 installed from Debian 10 repository. Best regards, Leonid Fainshtein On Thu, Aug 27, 2020 at 3:01 PM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Thu, Aug 27, 2020 at 8:58 AM Leonid Fainshtein < >

On Fri, Jul 3, 2020 at 3:32 AM Dovid Bender <dovid at telecurve.com> wrote: > > > On Mon, Jun 29, 2020 at 6:46 AM Joshua C. Colp <jcolp at sangoma.com> wrote: > >> On Sun, Jun 28, 2020 at 2:26 PM Dovid Bender <dovid at telecurve.com> wrote: >> >>> Hi, >>> >>> We have a box up and we are starting to see a lot of

Thanks for the info, Joshua. Does PJSIP handle database access the same way Chan_sip did? We had a number of boxes running chan_sip referencing the same mysql server without issue. We're going to attempt to get a backtrace on the next occurance. We're also going to run a local copy of the database on the same physical asterisk instance and have the system reference it. Just to

Hello! https://forums.asterisk.org/ is doing it again - "Content Encoding Error. An error occurred during a connection to forums.asterisk.org. Please contact the website owners to inform them of this problem". Which is odd, as the Qualys test seems to pass, only losing a point for supporting TLS 1.0. But I know it's not just me because Pingdom can't read the page, either.

Thanks Joshua; Working now - but regarding the slow wiki, it's normally 5-10 second page load for me in the UK, but it was closer to 20 seconds earlier. https://tools.pingdom.com/#5c862c04ad800000 Seems to be back to around 5 seconds now, but I notice that if I run a Pingdom page test from a US server, the page loads in about 1.3 seconds. I wonder if there might be a misconfigured edge

Hello, it did not seem the call hung. It seemed it never started. There was no dialplan execution on the asterisk side. It looked like phones were unregistered. Same shows the log posted previously. Marek Sent with Proton Mail secure email. ------- Original Message ------- On Wednesday, November 8th, 2023 at 1:21, John Harragin <jharragin at mw.k12.ny.us> wrote: > Marek, >

Could you show the phone configurations - section "Proxy and Registration" On Mon, 6 Nov 2023 at 23:13, Marek Greško <marek.gresko at protonmail.com> wrote: > Hello, > > you are probably right. It should somehow be related to DNS. I just found > out this in the storm of previous messages: > > WARNING[13945] taskprocessor.c: The 'dns_system_resolver_tp'

On Tue, Oct 27, 2020 at 5:35 AM Olivier <oza.4h07 at gmail.com> wrote: > Hello, > > Where can I find doc about PJSIP's ice_support parameter ? > > Do you need to configure things elsewhere in Asterisk config files > (rtp.conf, PJSIP transport sections, ...) to make ICE work properly ? > It needs to also be enabled in rtp.conf. > I'm asking because, if

Hi All This sounds just like a problem I have had and still investigating having moved to 16.9 using chan_sip. I am still trying to repeat the problem it looks from debug that the issue is either voicemail of call transfer but I cant consistently repeat it. Voicemail is using ODBC and I just imported the data from the old system into the new database. Nick - if you have any more info I

Sorry, should have included that. Asterisk 16.2.1 Mark. On Thu, 4 Apr 2019 at 14:56, Joshua C. Colp <jcolp at digium.com> wrote: > On Thu, Apr 4, 2019, at 10:53 AM, Mark Farmer wrote: > > As I understand it, delays like this are almost always caused by slow > > or failing DNS lookups. Running a packet capture on all interfaces > > filtering on port 53 shows no DNS