similar to: pjsip: occasional sip_transactio Unable to register REGISTER transaction (key exists)

On Wed, Oct 21, 2020 at 7:46 AM Michael Maier <m1278468 at mailbox.org> wrote: > Hello! > > On 20.10.20 at 14:00 Asterisk Development Team wrote: > > The Asterisk Development Team would like to announce the release of > Asterisk 18.0.0. > > This release is available for immediate download at > > https://downloads.asterisk.org/pub/telephony/asterisk > > I

On 13.07.20 at 00:17 Joshua C. Colp wrote: > On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins <dan at nimblea.pe> wrote: > >> Asterisk 18 will have support based on this asterisk update Matt F did for >> CommCon's sponsor slots >> >> https://youtu.be/eas1csaX-wc >> >> > As well support will go into Asterisk 16 and 17 as well. It's just been

On 13.07.20 at 10:54 Joshua C. Colp wrote: > On Sun, Jul 12, 2020 at 11:37 PM Michael Maier <m1278468 at mailbox.org> wrote: >> One more question, >> what about the pjsip pcap support? Will it be backported to Asterisk 16, >> too? Would be absolutely cool! Debugging encrypted SIP is really a pain. >> > > It can't be backported ... because it already is.

Hi Carsten, On 02/11/2018 at 07:46 PM Carsten Bock wrote: > Hi, > > Lost percent (%).... Are you sure? I'm seeing here: ...........Receive......... .........Transmit.......... Count Lost Pct Jitter Count Lost Pct Jitter RTT.... 188K 0 0 0.000 188K 16641K 8809 0.000 0.026 => This doesn't sound reliable to me: there are 188K packets and 16641K

On 17.05.20 at 01:28 Joshua C. Colp wrote: > On Sat, May 16, 2020 at 10:58 AM Michael Maier <m1278468 at mailbox.org> wrote: > >> => How are the RTT values exactly calculated? Which values are actually >> used for? >> > > The value is calculated according to the logic in the RFC[1]. Specifically > using embedded timestamps in the RTCP packets and

Hello, could somebody please tell me the meaning of "Pct" as seen in asterisk cli: ...........Receive......... .........Transmit.......... Count Lost Pct Jitter Count Lost Pct Jitter RTT.... Thanks, Michael

Could this gap in sequence numbers caused by a codec change generate errors like the one below? [2018-02-13 12:57:43] WARNING[4917][C-0004c2cb] codec_sangoma.c: [526559][g722toulaw] Got Seq 15944 but expecting 10106 (time since last read = 0ms), dropped 5838 packets On 02/13/2018 01:24 PM, Andres wrote: > On 2/13/18 11:55 AM, Michael Maier wrote: >> On 02/13/2018 at 08:41 AM Floimair

On 15.05.20 at 14:31 Doug Lytle wrote: > Google says Round Trip Time > > https://www.voip-info.org/asterisk-rtcp/ That doesn't answer my question (I know the abbreviation RTT). Therefore I'm trying again: I'm just wondering what the RTT *exactly* means. Where are the exact measuring points located? => How are the RTT values exactly calculated? Which values are actually

On 02/13/2018 at 08:41 AM Floimair Florian wrote: > No you're reading it wrong. > > There are 188K received with no loss, and 16441K transmitted. This doesn't make any sense to me, either. There can't be more packages transmitted than received. It's the same codec in and out and it's been running exactly the same time. > ...........Receive.........

/etc/dovecot/virtual/Inbox2/dovecot-virtual * -Trash -Trash/* -Deleted* -Junk* inthread x-references2 x-mailbox INBOX Using php to connect o {host}virtual/Inbox2 gives me [SERVERBUG] BUG: Unknown internal error. If I connect to virtual/Allmail which has "all" instead of the inthread line it works.

Hello, I finally got an INBOX to work, but is virtual mailbox plugin only meant to create a virtual inbox, or should I be able to have an Allmails virtual folder that is viewable in my imap client? if ~/virtual/ is my location, would I create an Allmails directory in the virtual/ or virtual/INBOX/Allmails. Im fairly confused on how this is suppose to work. I was trying to duplicate the gmail-like

Hello! Today, I tried to switch from asterisk 13.7.2 to 13.9, but I'm getting strange problem w/ the registering of all of my extensions. It looks like that: [2016-05-12 08:59:38] VERBOSE[2332] res_pjsip/pjsip_configuration.c: Contact 107/sip:107 at 192.168.15.73:5060 has been created [2016-05-12 08:59:38] VERBOSE[2272] res_pjsip_registrar.c: Added contact 'sip:107 at

hi, I am trying to add xenbus watch support to mini-os, but am getting into a endless loop of watches firing over and over. Below is an excerpt from the xenstore tracefile: IN 0x805acc8 20070103 19:32:18 WATCH (/local/domain/0/backend/vif/26/0/state 00043818 ) CREATE watch 0x805a8b8 OUT 0x805acc8 20070103 19:32:18 WATCH (OK ) OUT 0x805acc8 20070103 19:32:18 WATCH_EVENT

We have a customer where their VM running Asterisk appears to have crashed. Fortunately, we had some debugging enabled. The asterisk messages file has this... (in notepad+ the blank line in the middle is all [NUL][NUL] [NUL][NUL]....) [08/12 15:30:55.880] VERBOSE[6920] app_mixmonitor.c: Begin MixMonitor Recording CBRec/IS__a37ae004-c780-4c7f-88a9-a04402f0ab4e-0000e70f [08/12 15:30:55.881]

Hello all! I can see a strange problem during invite in dialog in the context of timer handling. Given is the following incoming call from provider at 8.195.88.234 (2 at 2) to my asterisk at 28.19.57.152 (1 at 1): After 900s suddenly *asterisk* starts the timer reinvite - I would have expected the reinvite started by the provider as usual. The expected reinvite by the provider is started

I've more-or-less successfully installed IE6 under 20050111. It works fine but PNG's don't display, so I tried reregistering the pngfilt dll: wine regsvr32 c:\\windows\\system\\pngfilt.dll wine: Unhandled exception (thread 0009), starting debugger... Usage: winedbg [--auto] [--gdb] cmdline (not sure why the debugger doesn't start properly. I did have debugging enabled when I

As any other daemon, I''d assume it should be possible to restart xenstored (i.e. if it crashed or if, during development, a fix was made to it that needs to be tested). However, as I learned after over a day of digging, things don''t work anymore after doing so: Since the watches are transient, they''re lost with the daemon going away. Right now I specifically need this in

I have been reading a number of the past threads about G.729 licensing., about how the registration keys are linked to the network configurations, limited number of registrations etc, etc. Is there no reason why the decoding can't be done in with some Asterisk compatible hardware, so that once the adapter is bought, all licensing issues go away. In that way the owner could fiddle with the

This may not strictly be an asterisk question, but not sure where else to post ... I have an Asterisk test server setup with two firefly clients, one on the local lan and one on an external ip address. Both clients are setup the same way and voice calls work fine. The asterisk console reports a "Registered" message for the external client at about one minute intervals but the

I start asterisk 13.7.2 and it dies before I can rasterisk into it. I've tried getting a coredump, but it doesn't coredump. I know there are a lot of errors in the log below, but most of those just say it'll not load a module, and no big deal. When launching from commandline (not service script) here is what happens. http://pastebin.com/3GFe6fG9 Travis Ryan Director of