Re: [asterisk-users] Nagios asterisk monitoring
Brandon,
I wanted to show my support this module as well. I would appreciate
information on how to obtain the finished product and or help for beta
testing.
Kenny
On Wed, 2007-04-11 at 14:11 -0500, Brandon Kruse wrote:
> I wrote a very extensive plugin for cacti to monitor asterisk.
>
> It uses the manager interface to poll and get statistics for 1.4 and 1.2.
>
> Let me know if you interested, ill post it, or email me directly.
>
> -bkruse
>
>
> voip crazy wrote:
> > Dear list,
> >
> >
> > I am trying to configure the nagios plugin called check_sip. I just
> > read the README file included with the plugin. I follow the readme
> > steps to configure the plugin, without success. In the nagios web
> > interface I can see (No output!) In the status information column. If
> > I run the chech_sip plugin from a linux console, I get
> > /usr/local/nagios/libexec# ./check_sip -u sip:[EMAIL PROTECTED]
> > SIP 200 OK: 0.00 second response time
> >
> > I do not know why If I run the plugin from the consle it works ok, but
> > if I run it from Nagios web interface it does not run.
> >
> > Anyone are using this plugin?
> > Could you helpme to solve that?
> > Any clue will be appreciated.
> >
> > Thanks for your time.
> >
> > VoipCrazy
> >
> > Here goes my nagios check_sip plugin configuration.
> >
> > define command{
> >command_namecheck_sip
> >command_line$USER1$/check_sip -u $ARG1$ -H $HOSTADDRESS$ -w 5
> >}
> >
> >
> > define service{
> >use generic-service
> >host_name -PBX
> >service_description SIP test
> >check_command check_sip!sip:[EMAIL PROTECTED]
> >contact_groups admins
> >max_check_attempts 4
> >normal_check_interval 5
> >retry_check_interval1
> >notification_interval 240
> >check_period24x7
> >notification_period 24x7
> >notification_optionsc,r
> >}
> >
> >
> >
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[asterisk-users] Viable using purchasing sip lines
Hello All, We have been doing Asterisk and CME implementations recently but we almost always exlusively bring in analog lines and or PRI for PSTN access to our systems. I have known about providers providing SIP based lines and SIP trunks to end users for PSTN access. I am curious about the following: - How practical is this? The idea of terminating pstn calls to across the Internet which is an unguarenteed medium concerns me. Even if our access to it is quazi stable T1 data type of access. Do any of you do systems where this is soley the method used for incoming calls from the pstn? If this is done are there things to look for in a SIP provider, as in their presence on the Internet latency ..etc? - What are the major advantages? I know some places provide all you can eat plans which could be seen as a plus and some others provide really low rates. Are there others? - Who are the major players? How are these usually ordered and identified? - Any general tips? Thanks all! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quantum Voice Asterisk?
Are there any subscribers to Quantum voice on here who have gotten Asterisk to work with their service? I found a small how to for [EMAIL PROTECTED] but have still not gotten it to work. I am registering with their sip fine but am having a hard time passing calls to them. Kenny ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AstriCon 2006 Location
The best place for Astri Con 2006 would definatly be Omaha, Nebraska! ;) very central ...ah one could hope. __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple Line Appearances / Why use this?
I apologize for the double post. I am curious as to what the usefullness is of the multiple line appearance feature on Polycom phones. I setup our phones to register one line per extension but I hear the IP501's can do three line appearances. Why and how could this feature be applied? Thanks again all. Kenny __ Click here to donate to the Hurricane Katrina relief effort. http://store.yahoo.com/redcross-donate3/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Server Brand
Hello, all I apologize for all these questions. But I have so many running through my head for this. What brand of server is a good one to use for running an asterisk box? I did an install the other day on a Compaq Proliant ML370 G2 / Debian Sarge and it is currently working great. But HP/Compaq costs bukooo bucks. I have heard Super Micro is what alot of you are using. Any problem running Debian on this? If I am correct on this what makes this brand / their product so appealing? Just price point? I would love to use a 1U server but for I need more that one PCI port :) (at least three) for the TDM cards. Any thoughts are always appreciated. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Good Polycom Dealer?
Could any of you provide me information on a good Polycom phone dealers to utilize. One who provides firmwares ..etc Thank you! Kenny __ Click here to donate to the Hurricane Katrina relief effort. http://store.yahoo.com/redcross-donate3/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel Not Sending Tones
Hello everyone, I had an asterisk box which was
working great bt now for some reason I cannot dial out
on any of my outside lines. I am using a TDM card
with 4 FXO ports.
System:
Debian Sarge, 2.6.8-2-386
Compaq Proliant ML370 G2 Server
Polycom IP500 Phones
dtmfmode=rfc2833
I initially set up my system by comiling the latest
cvs zaptel drivers and then the lastest cvs asterisk.
Configured my system and every thing seemed to worked
great. But somehow I changed something or my TDM card
is toasted because now when I dial an outside line I
always receive the telecom standard recording that my
"call could not be completed try again"
I am using the following extensions.conf line to dial
an outside line:
"exten => _9X.,1,Dial,Zap/g1/1-4/${EXTEN:1}"
When I dial a number such as 9555 on my phone
asterisk CLI shows this:
-- Executing Dial("SIP/202-115a", "Zap/g1/1
4/555") in new stack
-- Called g1/1-4/555
-- Zap/1-1 answered SIP/202-115a
Then I hear afer about 7 seconds the operator message.
I have listened in with an analog handset to what is
happening on the line when asterisk runs this command
is sending tones... but they seem to be not "spaced"
correctly.
- I tried switching from koolstart signaling to
loopstart no change.
- I tried installing the Sarge zaptel package of
drivers and then recompiling asterisk. No change.
- I tried recompiling latest cvs of zaptel and
asterisk, no change.
I have the following setup in my zaptel.conf
loadzone = us
defaultzone=us
fxsks=1-4 (switched this to fxsls for loopstart
testing no change)
zapata.conf
[trunkgroups]
[channels]
context=default
switchtype=national
signalling=fxo_ls
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=no
restrictcid=yes
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
group=1
context=default
channel => 1-4
To conclude, I have tested this on different physical
phone lines, different extensions, different channels
on the tdm card and the same result happens. Tested
the phone line without asterisk... no problem lines
work fine?
Any help ?
Thanks all
Kenny
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[Asterisk-Users] Polycom IP500 / Registration Question?
Hello again, I have a bunch of Polycom IP500 Phones with Boot 2.6.2 and SIP 1.4.1. I have defined seperate user and peer settings for my extensions as per posts I have seen in here. I can access voicemail...etc and the phone seem work fine. Question: when I do "sip show registration" there is nothing listed and/or "sip show subscriptions" nothing is there. But when I do "sip show peers" I see a list of my phones, same for "sip show users". Shouldnt I see my phones as "registered" or something similar under these two sections? I have them set to register, and like I said they are working fine. Any help? Thanks, Kenny __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbiew extensions.conf question
I am newbie trying to setup about 12 Polycom Ip500's on an asterisk server. I am working on my extensions.conf and am trying to make it so that all my extensions can dial each other. My extensions are number 720, 721, 722, 723 ..etc in my from-sip context I began doing entries such as: exten => 720,1,Dial(SIP/720,20) exten => 720,2,Voicemail(u720) exten => 721,1,Dial(SIP/721,20) exten => 721,2,Voicemail(u721) ..etc ..etc This is not a big deal for such a small number of extensions but I was thinking about larger installs.. this would begin to suck. Is there anyway around this? Thanks! Kenny Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
