Re: [asterisk-users] nagios asterisk check SIP

On Fri, Jun 17, 2016 at 11:22:48AM +0200, Thomas wrote: > Iam loocking for an programm to check the SIP port of an Asterisk asterisk. > > Ome time ago I have used > #/usr/bin/sipsak > but it seemed that it is not working anymore? Hi Thomas, Maybe this links help you:

Re: [asterisk-users] SPA112 flapping

Hi Mike, I would try the following: * If you can login through HTTP, check the uptime of the Cisco device. Make sure the device is not rebooting. * If you can, make a 'ping' from the PBX to the device and annotate milli-seconds of response. Then compare then to the default 'qualify' sip setting

[asterisk-users] Compiler errors when 'make asterisk' for D100 transcoding board

Hi there ! Someone in this wonderful list tried to install Sangoma transcoding board D100 on Asterisk v11 ? I followed each of the steps in the wiki [1], but when running 'make asterisk' receipt compilation errors about the absence of some header files [2]. I exchanged some mail with the

Re: [asterisk-users] queue_log - odbc vs AMI

Hi Marek, Here, we have an Asterisk v11-cert11 and found that there is NOT equal the CDR via AMI and CDR in Database. Please, check my gist: https://gist.github.com/MefhigosetH/89462e599a996dedf048f8d2b4e94d47 We have in use some custom dialplan variables in CDR (ie.: groupcount and rptqos),

[asterisk-users] Compiler errors when 'make asterisk' for D100 transcoding board

Hi List, I solve this issue and I want share it with this community. The sng-tc-linux-1.3.8 package don't compile across Certified Asterisk. Only normal Asterisk like 11.22.0 version. We have this version in production with the D100 board. Working. Cheers -- GnuPG Key ID: 0x39BCA9D8

[asterisk-users] Identify more demanding routine inside Asterisk

Hi List ! I'm facing a problem with the CPU consumption in Asterisk 11.22.0. I could decrease a lot of load, migrating both the astdb.sqlite3 and call recordings (with Monitor app) to a tmpfs mount in RAM (with noatime and nodiratime flags), manually spread each of the hardware interrupts

Re: [asterisk-users] how to read sip debug

Hi Thufir, The analysis of a SIP Debug depends on what the problem to be solved. If you experience problems with inbound calls from a SIP trunk or provider, you can type in Asterisk cli 'core set debug 3' and then 'sip set debug ip xxx.xxx.xxx.xxx' where xxx is the IP of your SIP provider or

Re: [asterisk-users] Using g729 now that patents have expired

Hi Steve, I understand your question and your point, but I use the g729 codec from the link that Carlos share, for almost 6 years from Asterisk 1.4 to v13 without a single problem. So, sory but I don't share your phrase "from a lesser know web site". About your question, I did not known that

Re: [asterisk-users] Which tool to automatically restart Asterisk ?

Hi, Oliver. Maybe something like this (add this script to your crontab): 8<-- #!/bin/bash # # File: asterisk-watchdog.sh # Date: 2015.05.26 # Build:v1.0 # Brief:Secuencia para monitorizar procesos. # # ${PATH}:

Re: [asterisk-users] Turn on SIP debugging from DialPlan

Hi Derek, SIP debug can be enabled via Asterisk CLI (console) with the command: asterisk> sip set debug on If you know via what trunk your call goes, you can use the following command instead: asterisk> sip set debug ip xxx.xxx.xxx.xxx Where the xxx is the IP of your trunk (voip to pstn

Re: [asterisk-users] Disallow CALLS without registry

Hi Antony, Sory but I don't understand why your Asterisk accept anon calls with the conf you provide us. Maybe a full excerpt of an incoming call will help. Last, there exist dialplan like GROUP and GROUP_COUNT that permits you count the number of calls in a custom group fashion. El 10/2/2017

[asterisk-users] Asterisk 11.23 with libmysqlclient20 on Debian 8

Hi List! I'm facing a problem while compiling Asterisk-11 on a Debian 8 server. The mysql-server version installed is 5.7 and come from the official mySQL community repo for Debian. After compile, install and execute Asterisk, the comman "lsof -p `pidof asterisk` | grep mysql" don't produce any

Re: [asterisk-users] asterisk-users Digest, Vol 147, Issue 5

Hi all ! Thanks for your feedback and sory for the delay. Respond: > Date: Mon, 3 Oct 2016 21:05:55 -0300 > From: Marcelo Terres > > I think that you need the dev files too. In Debian 8, the package is > libmysqlclient-dev. > > But Debian 8 uses libmysqlclient-18. Where did

Re: [asterisk-users] Asterisk 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3 : freeze on 'sip reload'

Hi Jonas! Do you currently use any TLS technology in your Asterisk? Like SIP-TLS o pjSIP-TLS support ? If don't, please go to modules.conf and start disabling some modules that you don't use. For example, I can see some other modules related to calendars. If you don't use this, please disable

Re: [asterisk-users] iowait issues on CentOS 7

Hi Luca, IO delay maybe come from Hard Disk lattency. You can exec an "lsof " command to view what file asterisk proccess hold down when load spike. If there are some call recording, you can configure Asterisk to make it in a temp location, a RAM Disk in Linux. If you make hard usage of the

Re: [asterisk-users] iaxmodem errors.

Hi John! I'm not sure why are you using iaxmodem... I use it a few years ago with Asterisk 1.4 In Asterisk v11 fax is managed using res_fax. Please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_ReceiveFAX_res_fax You only need download, compile and install the spandsp

Re: [asterisk-users] Force hangup not working on stuck channel

Hi Carlos, Did you try with the following CLI command: CLI> channel request hangup CHANNEL_NAME ??? El nov. 3, 2016 1:16 PM, "Carlos Chavez" escribió: > I am unable to force a hangup on a channel that has been stuck for over > two days: > > IAX2/from-CD-11006

Re: [asterisk-users] send a call to moh until user is available

Hi Tux John, The behavior you need is cover in Asterisk within a Queue. 1. Create a new queue in queues.conf and assign as static member, the 4450 extension. 2. In your dialplan, you need to route the incomming calls to the new queue and pass as argument the timeout in seconds you want when

Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit

Hi Motty, Please, set Verbose to 3 and Debug to 3 At Asterisk CLI. Then "sip set debug on". Now try to register again. At last, " sip de debug off". Examine tour console or full log file to find some clue ir send me back some trace. Cheers. El oct. 13, 2016 1:45 PM, "Motty Cruz"

Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit

Ok. Please, note that 192.168.1.37 (I suspect) is the internal LAN address Of the Polycom hardphone. If this is true, then you have NAT issues. The REGISTER message are received by your PBX, but when respond, Asterisk send the next SIP message to the IP informed by the phone, that is the

Re: [asterisk-users] Asterisk - Vtiger integration

Hi Alejandro, The documentation about your question is here: https://wiki.vtiger.com/vtiger6/index.php/PBX_Manager After a few seconds of read, I think that VTigerAsteriskConnector can run on a separate server than Asterisk PBX. VTigerAsteriskConnector connects to Asterisk via Asterisk Manager

Re: [asterisk-users] Cisco IP 8841 asterisk integration

With all the money you plan to invest in firmware, licenses, etc., you have bought a Grandstream IP phone or Yealink... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

Hi Yves, Maybe your switch put your Polycom inside a Voice VLAN, based on the MAC of the phone. Maybe with the snom this not happen because your switch don't see the MAC of the Snom as a "supperted IP Phone". 2016-12-21 13:59 GMT-03:00 Yves : > sorry... typo > the

Re: [asterisk-users] Manager events showing in CLI

Hi Ron, I don't remember right now, but you can try this command: cli> manager set debug off Cheers El 26 mar. 2017 3:58, "Telium Technical Support" escribió: I somehow cause AMI events to appear as output in the CLI, and I can’t figure out how to turn them off. Can

Re: [asterisk-users] Manager events showing in CLI

Ok, Please, check your manager.conf and logger.conf for any clue about debugging options, into the Asterisk configuration directory. El 26 mar. 2017 14:52, "Telium Technical Support" escribió: > I tried that but it had no effect. Still see things like: > > > > [2017-03-26

Re: [asterisk-users] PBX selection

Hi Speed Boy. I agree with Emiliano Vazquez too. Additionally, you and your team must think others points before choose Asterisk: * Asterisk is build to work on Linux. So your team needs some skills like setting up a basic Linux server (Debian, Centos, etc), donwload software from Internet,

Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

Hi, Jerry, I don't know what S.O. you have in the Server, but you can check the man page (https://linux.die.net/man/8/in.tftpd) for tftpd and use the options --address, so you can tell tftp from what interface/port this service listen request. >From the IP in your logs (69.64.57.18) the request

Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

Hi David, Tim, Try to use Bail2Ban at last resort. Fail2Ban is a ractive approach, that permit the traffinc AND ONLY BLOCK them after certain level triggered. Use iptables to block the unused services faced to public networks like Internet. And configure these services properly, so they listen

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

Hi Ernie, When one-way audio appear (no matters if there is a VPN or NAT server on the diagram) I simply : * Enable SIP debug on Asterisk server. Excecute 'sip set debug ip x.x.x.x' on Astrisk CLI, where x.x.x.x is the IP of the phone or SIP peer you want to debug. * Make a test call and

Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

Hi Nathan, Personally, I create a git repo on /etc/asterisk/ folder. With this approach, you not only can backup current dilplan on another location (another private server, or private repo on Bitbucket account). You can follow all the change history you made. Simply install git, then go to

Re: [asterisk-users] Voicemail asking for login

Hi Darcy, What Pete think is correct. Maybe excecuting the following command at Asterisk console, will help you: asterisk> voicemail show users And you will get a list of all mailbox configured in your system. Search for the user with problems. Finally, in the Asterisk wiki you can find more

Re: [asterisk-users] Automatically dial a number, then an extension

Hi John, I think we need to known how you play the audio to the customers, before we can help you. Are you using AMI? Or AGI maybe? Or Call files? What Asterisk version do you have? El 15 may. 2017 12:35, "Tech Support" escribió: > All; > > I have an application

Re: [asterisk-users] IAX port 4569

No. The 0.0.0.0 listen address is fine. El 5 jun. 2017 10:06, escribió: > I'm getting: > netstat -a |grep 4569 > udp0 0 0.0.0.0:45690.0.0.0:* > > Should I be getting localhost IP? > > Thelma > > On 06/05/2017 06:48 AM, the...@sys-concept.com

Re: [asterisk-users] IAX port 4569

Dear Thelma, Yes. Asterisk listen on port 4569 UDP on default config. Please, look at the Asterisk logfile, for clues about your issue. Or enable IAX2 debug vía Asterisk CLI. Other ideas: * Check that your server firewall permit UDP port 4569 incoming traffic. * Run tcpdump over the network

Re: [asterisk-users] IAX port 4569

Another idea: * Run netstat -tulpn command on Linux box AND look if there are an Asterisk process listening on 4569 UDP port on 0.0.0.0 El 5 jun. 2017 10:00, "Victor Villarreal" <mefhigos...@gmail.com> escribió: > Dear Thelma, > > Yes. Asterisk listen on port 4

Re: [asterisk-users] IAX port 4569

I think you need to increase verbose output and search in /var/log/asterisk/full for any error message related to IAX2 registration or simil. 2017-06-05 17:12 GMT-03:00 : > No, I don't think it is IP table issue, I've not upgraded dd-wrt for a > while and it was zoiper

Re: [asterisk-users] *****SPAM***** Re: IAX port 4569

my port is 4569 is in Stealth mode (so it is closed) :-/ > > > Thelma > On 06/05/2017 02:19 PM, Victor Villarreal wrote: > > I think you need to increase verbose output and search in > > /var/log/asterisk/full for any error message related to IAX2 registration > >