similar to: Can't hangup channel from CLI

Hello All, I'm using Asterisk 16.8.0 on a Centos 7 box. Previously 16.5.0, But recently upgraded to attempt to resolve this issue. Using bundled PJSIP. The PBX is using mysql realtime for most functions. The Mysql server is on the same lan as the asterisk box. As more users have been moved to this box. It's become unstable. Randomly, I'll start seeing "WARNING[12667]

Hello, I work for a company that is experimenting with the implementation of Asterisk. We have a VoIP provider that is giving us a demo account with 200 minutes on it. We can register with their service but cannot complete calls with Asterisk. We can use a Grandstream GXP-2000 with the supplied registration info and it does work. We can also register Asterisk with FWD and dial the FWD

On Fri, Feb 17, 2017 at 5:17 AM, Olivier <oza.4h07 at gmail.com> wrote: > Hi George, > > How does ast_coredumper compare to ast_grab_core ) ? > Is it worth learning to use both or shall favor one ? > > PS: As I don't know either program, yet, my question may seem silly. > Please, forgive me for this > Not silly at all. ast_grab_core actually kills asterisk to

Hello, I'd like to know if anyone of you is finding my same problems using any recent asterisk version, after 13.7 / 13.8 with chan_sip. If I use any recent asterisk version, after just few seconds asterisk completely locks up, stopping processing SIP/UDP packets. Nothing is written in the asterisk log, but if I run "netstat -nap | grep 5060" I see the UDP buffer filled up. If I

Hello list, I've got a whole lot of these in my debug log : [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both fail to provide 160 samples [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Failed to get 160 samples from read factory 0x1cea33a0 [Apr 18 15:12:44] DEBUG[26973] audiohook.c: Read factory 0x1cea33a0 and write factory 0x1cea3dd8 both

Hello I am experiencing a freeze of the Asterisk proces when issuing a 'sip reload'. I have this issue every time on asterisk versions : 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3. I do not have this on versions certified-13.8-cert2, certified-13.8-cert1 and asterisk 1.8.32.3. The only solution is a cold restart of Asterisk. I can execute any command on CLI except 'sip

Hi, We have a box up and we are starting to see a lot of "Exceptionally long queue length queuing" in the logs. From all the research so far it seems like this leads to their systems crashing and being unreachable. In our case the box remains up and takes calls. We are running Asterisk 16.6.1. We are using MusicOnHold to play online music streams via ffmpeg. Any idea on how to

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 "Exceptionally long >> queue length queuing" in the logs. From all the research so far it seems >> like this leads to

In the past, there have been reports of problems with Asterisk with multiple processors and/or HyperThreading. I'm having a !@#$ of a problem with an HPDL380 with 2 3.4gHz Xeon processors, 2 gb RAM -- if I got 24 hours I'd think I had died and gone to heaven :) Am I missing something obvious like "Asterisk is single CPU, single core?" I can't access the ILO so I

On Mon, Mar 2, 2020 at 2:52 PM Nick Olsen <nick at floridavirtualsolutions.com> wrote: > Hello All, > I'm using Asterisk 16.8.0 on a Centos 7 box. Previously 16.5.0, But > recently upgraded to attempt to resolve this issue. Using bundled PJSIP. > The PBX is using mysql realtime for most functions. The Mysql server is on > the same lan as the asterisk box. > > As

Hi list, I have to connect two asterisk box, in this scenario: [asterisk1]----sip----[asterisk2]----PSTN I must use sip, cos we'll use cisco rtp header-compression to save bandwidth. Could you tell me the best way to send calls from asterisk1 to asterisk2, since I cannot use IAX trunking? Thanks in advance Eduardo

This just started after upgrading to 11.11.0. After a call is completed (both ends hang up) the call still shows as active. # asterisk -x "core show channels" Channel Location State Application(Data) SIP/thinktel-0000000 (None) Up AppDial((Outgoing Line)) SIP/4164251212-00000 4165555555 at LocalSets Up Dial(SIP/thinktel/4165559999) 2 active

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

I am using a cyberdata "sip paging adapter" and with the Dial(MulticastRTP/basic/IP:port) and with tshark I see the RTP data, my device looks like its accepting the data and I hear a click for my relay on my device so it would seem its accepting the call, however - I hear no audio... Asterisk 11.11.0 is what I am using. What might be wrong here? Thanks, jerry -------------- next part

I am seeing this in my log file :[Aug 7 21:35:24] ERROR[19582] acl.c: Cannot create socket [Aug 7 21:35:24] WARNING[19582][C-00000283] res_rtp_asterisk.c: Unable to allocate RTP socket: Too many open files [Aug 7 21:35:24] NOTICE[19582][C-00000283] chan_sip.c: Failed to authenticate device "677"<sip:677 at IP>;tag=3637370132313231383238343335 [Aug 7 21:35:24] WARNING[19734]

Running 11.10.2 on NetBSD 6.1.4 but I observed this on 11.11.0 as well. I have a directory which, through a combination of NULL mount and UNION mount contains everything in the installed config directory /usr/pkg/etc/asterisk except for my modified versions of those files. Basically I mount_null /usr/pkg/etc/asterisk on /usr/local/etc/asterisk and then mount_union my SVN directory with my

All, I have successfully executed the "deploy" task in Capistrano/Switchtower to establish a symbolic link named "current" which points to the current version of my app. on my remote server. My app. was already deployed to the existing Rails root directory (call it "appname") though. I have Apache fronting my app. through a virtual host whose doc. root is itself

Greetings- As many of your are Polycom "experienced", I was hoping some kind soul could provide direction on a specific issue. On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding an instance where, using intercom/paging functionality of FreePBX, I need to override an end user's 'Do Not Disturb' selection on the handset. By default, DND simply

Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer. There's probably something I've changed that causes this

Issue is what subject says. Here is the background. Version: 11.11.0 Topology: Asterisk Box at our Data Center behind Cisco Firewall. Everything works fine from remote offices over a VPN. Issue is sales team would like to connect up to our Asterisk box remotely (offnet). Common enough solution, I'm guessing. So, I've opened all the correct holes on the firewall and hammered out