similar to: SIP behind NAT (*sigh*)

Hi I have a brand new ATA186 with the following firmware: Version: v2.15.ms ata186 (Build 020919a) I have been through the archives about how to configure it, but my colorful configuration web page does not have the same fields that people say I need to adjust. Even the examples on Cisco's web site don;t match. For example, I don't have the GtkOrProxy field, which is an important

I noticed a few issues with app_queue just wanted to know if its sip related or ata186 related: Ext 111 and Ext 112 are dynamically loged into the queue via AddQueueMember. Call hits queue with fewestcalls routing. Rings ext 111 if 111 doesn't answer. It rings ext 112. If for some reason ext 112 doesn't answer it rings back to 111. Again at this point ext 111 isn't answered it

I dont have a cisco acount yet can some bady hel me with the ata18x-v2-16-030401a-1.zip file. thanks in advance Rodney Acosta Coya. Dpto. Tecnologia de la Informacion. racosta@moanickel.com.cu Tel:(53)(24) 62 611 -----Mensaje original----- De: Paul Rodan [mailto:asterisk@glitch.cc] Enviado el: Martes, 23 de Noviembre de 2004 11:24 a.m. Para: 'Asterisk Users Mailing List - Non-Commercial

Hi all: I've got some core dumps with chan_skinny. The client is ATA186 with v2.16.1.ms ata18x (Build 030814a). The * version is CVS-10/05/03-16:03:26. When I make a call, the phone connected with ATA rings only 1 time and * dies. Maybe I have some errores in ATA config. If someone has proven configs for ATA, please send me the details. Thanks in advance, Gus The logs: *CLI> Version

Hi Eric, Thanks for your reply. I do have nrpe running under xinetd on the host I'm trying to monitor. And running the nrpe checl locally: [root at ops:~] #/usr/local/nagios/libexec/check_nrpe -H localhost NRPE v2.15 [root at ops:~] #grep only_from /etc/xinetd.d/nrpe only_from = 127.0.0.1 216.120.248.126 And I do have port 5666 open on the security group for this host.

I've installed Asterisk and configured an ATA-186 as described at this link: http://www.djernes.org/~shawn/ata186.htm Unfortunately this guide abruptly ends before it explains how to deal with the sip.conf and extensions.conf files. So I left extensions.conf alone and my sip.conf looks like this: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0

Hi, I'm having trouble getting caller*id to appear on my phone connected to an ATA186, and being called from Asterisk. Does anyone out there successfully see callerid on their ata186-connected phone? The "From:" header in the INVITE to the ATA seems to have the "right stuff" - eg From: "Study phone" <sip:6002@195.217.255.45:5062>;tag=as412db061 But

I strange error messages when using mgcp and ata186 . This session is simply dial into 600 demo extension - echo test ... Handling request 'NTFY' on aaln/1@10.0.1.19 Transmitting: 200 29 OK to 10.0.1.19:2427 -- Endpoint 'aaln/1@10.0.1.19-1' observed '0' -- MGCP Asked to indicate tone: on aaln/1@10.0.1.19-1 in cxmode: sendrecv Posting Request: RQNT 306

Tim, where did you installed this nrpe package? is selinux running enforcing mode (getenforce command), try disabling with setenforce 0. why you are running it under xinetd as usual way is to run it as nrped daemon. test against with check_nrpe, not using telnet. -- Eero 2015-05-04 2:27 GMT+03:00 Stephen Harris <lists at spuddy.org>: > On Sun, May 03, 2015 at 07:23:19PM -0400, Tim

Hi, in following of a recent discussion I got to work on MGCP with the Cisco ATA186 again, and got it to work very nicely. However, there is a little thing with transfers I would like to get comments on: Call comes in from PSTN and goes to an ATA186 (MGCP) Call is answered and then, using flash, transferred to another extension If the extension is available, there can be an announcement and

> This is strange... > Do you have SSL aktive on both systems? Run nrpr localy without parameters > (this should return some nrpe stats) and check ldd for libssl. I don't seem to have that command. [root at monitor1:~] #find / -name "*nrpr" 2> /dev/null [root at monitor1:~] # And that's on either system. And if I do an ldd on both, this is what I can tell:

Am trying to install perl module "File::Find", but not able it gave the following cpan[1]> install File::Find CPAN: Storable loaded ok (v2.15) Going to read /root/.cpan/Metadata Database was generated on Thu, 21 Aug 2008 02:03:21 GMT Running install for module 'File::Find' The most recent version "1.12" of the module "File::Find" is part of the

Hello All, I am trying to get some ATA 188 units to register with my Asterisk box over SIP. I continue to get the same "401 Unauthorized" Error when they try to register. If I turn Sip registration off, I can use the phones without any problems with a static IP assigned in my sip.conf file, but I can't get the second phone port working. I've set up two separate logins both

Oh my mistake. I mean nrpe without parameters. It should say something about SSL/TLS aktiv or so. You could test nrpe without SSL. Use nrpe -n - H host Am 01.05.2015 13:18 schrieb "Eero Volotinen" <eero.volotinen at iki.fi>: > well. how about trying default setting and running nrped without xinetd. > > -- > Eero > > 2015-05-01 14:14 GMT+03:00 Tim Dunphy

Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz -------------- next part -------------- >

I'm trying to register with net2phone. I've already changed chan_sip.c, User-Agent: string to say "User-Agent: Cisco ATA 186 v2.16 ata18x (030401a)". But still I'm getting the error msg. Here is the debug msg: IP Address is xxx.xxx.xxx.xxx 11 headers, 0 lines Reliably Transmitting: REGISTER sip:66.33.146.12 SIP/2.0 Via: SIP/2.0/UDP

Hi NRPE: Error receiving data from daemon Seems as this is not a SSL Problem. Do you have a nagios user account? Cat /etc/passwd Am 01.05.2015 18:45 schrieb "Tim Dunphy" <bluethundr at gmail.com>: > > > > Oh my mistake. I mean nrpe without parameters. It should say something > > about SSL/TLS aktiv or so. > > You could test nrpe without SSL. Use nrpe -n -

> > It's listening on both IPv6 and IPv4. Specifically, why is that a problem? The central problem seems to be that the monitoring host can't hit nrpe on port 5666 UDP. [root at monitor1:~] #/usr/local/nagios/libexec/check_nrpe -H puppet.mydomain.com CHECK_NRPE: Socket timeout after 10 seconds. It is listening on the puppet host on port 5666 [root at puppet:~] #lsof -i :5666

Hello, I am trying to monitor a host in the Amazon EC2 cloud. Yet when I try to check NRPE from the monitoring host I am getting an SSL handshake error: [root at monitor1:~] #/usr/local/nagios/libexec/check_nrpe -H ops.jokefire.com CHECK_NRPE: Error - Could not complete SSL handshake. And if I telnet into the host on port 5666 to see if the FW port is open, the connection closes right away:

Hi all some how I'm getting confused: it seems I clobbered incoming calls from my sip provider. I can not imagine that my upgrade from 15.3 to 15.5 could be related I'm certain that dialling my own number, results in reaching asterisk, from my tcpdump. And on the asterisk console I get: pbx*CLI> == Using SIP RTP CoS mark 5 > 0x7f49ac54c040 -- Strict RTP learning after