69 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
> > or simil.
> >
> > 20
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 was working OK with my prev
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 wrote:
> > Does asterisk list
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" escribió:
> Dear Thelma,
>
> Yes. Asterisk listen on port 4569 UDP on default config.
>
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 in
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 that dials a list of numbe
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 o
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 c
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 replica
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 i
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, comp
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 /etc
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 13:49:39] DEBUG[2088]
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 someone offer a command w
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}: Varia
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 provid
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 1
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 the
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 I
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 problematic phone has the 1
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 communi
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 AstD
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
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 oficina 2770
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
intern
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" escribió
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 it.
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 hang
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 you get the 20 ?
>
>
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
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 from
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
https://w
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 (networ
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:
http://fabian-affolter.ch/blo
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), and
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 f
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 offici
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