[asterisk-users] system metrics to see if Asterisk is getting overloaded

[Nitesh Bansal]

Hello,

I'm trying to understand if I could use a system metric like load average,
cpu usage... to
decide if Asterisk is overloaded and if it is overloaded, I would like to
stop routing the traffic
to that box.

Is there any recommended system metric which you guys use to measure the
Asterisk load,
or may be is there any metric within Asterisk which could indicate that
system is slowing down?

Thanks,
Nitesh
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Re: [asterisk-users] VoIP monitoring tools

[Nitesh Bansal]

Hello,

Nagios is really a good tool, we are already using it to monitor Asterisk.
I'm looking for something which can monitor the Asterisk and store the
information
somewhere in a DB, where I could retrieve it from Kamailio.
I need that information in Kamailio to make routing decisions.

Thanks,
Nitesh

On Tue, Sep 27, 2016 at 4:28 PM, Tech Support <aster...@voipbusiness.us>
wrote:

> Hello;
>
> We’ve been using Nagios and a lot of customizations for the plugins
> for several years now to monitor over 1,000 metrics on each of our PBX’s.
> We’re in the process of GPL’ing the Asterisk plugins now. That gives us our
> core monitoring, notifications, event handlers, etc. To put it all
> together, instead of the standard static rrdtool graphs, which we thought
> would be a bottleneck and limit us, we’ve implemented a dynamic
> dashboarding system that we use to display the relevant data and to
> trivially create new dynamic dashboards in a matter of seconds. There is
> definitely no shortage of monitoring and NMS systems out there, but for us,
> we pretty much built this to monitor our Asterisk PBX’s. If you want to
> take a look at what we use, contact me offline.
>
> Regards;
>
> John V.
>
> supp...@voipbusiness.us
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] *On Behalf Of *Nitesh Bansal
> *Sent:* Tuesday, September 27, 2016 4:34 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
> Developers Mailing List
> *Subject:* [asterisk-users] VoIP monitoring tools
>
>
>
> Hello all,
>
> The question isn't directly related to Asterisk, but I'm looking for
> recommendations
>
> for a monitoring tool to monitor the health of Asterisk instances running
> in Production.
>
> Ideally, the tool should be able to generate monitoring traffic (OPTIONS
> ping or INVITE),
>
> use the response/no response from Asterisk to store the health of an
> Asterisk instance running
>
> somewhere in the DB.
>
>
>
> Thanks,
>
> Nitesh Bansal
>
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Re: [asterisk-users] VoIP monitoring tools

[Nitesh Bansal]

Thanks, I'm considering Homer, but I'm not sure if it can generate traffic
on its own to check
the health of the service.

Regards,
Nitesh

On Tue, Sep 27, 2016 at 11:17 AM, sysad...@reed-media.com <
sysad...@reed-media.com> wrote:

> Hello,
>
> you can have a look on Homer
>
> http://sipcapture.org/
>
> regards
>
>
>
> On 27/09/2016 10:39, Gholamreza Sabery wrote:
>
> Hello,
>
> For service monitoring you can use tools like sipsak in combination with
> Zabix or Zenoss. Also using Zenoss or Zabix you can monitor the health of
> your servers. This way you have both top-down and bottom-up monitoring. For
> monitoring call quality you can use tools like VoIP Monitor (it is not
> free).
>
> Regards
>
>
> On Tue, Sep 27, 2016 at 12:03 PM, Nitesh Bansal <nitesh.ban...@gmail.com>
> wrote:
>
>> Hello all,
>>
>> The question isn't directly related to Asterisk, but I'm looking for
>> recommendations
>> for a monitoring tool to monitor the health of Asterisk instances running
>> in Production.
>>
>> Ideally, the tool should be able to generate monitoring traffic (OPTIONS
>> ping or INVITE),
>> use the response/no response from Asterisk to store the health of an
>> Asterisk instance running
>> somewhere in the DB.
>>
>> Thanks,
>> Nitesh Bansal
>>
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>>   http://www.asterisk.org/community/astricon-user-conference
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
>
>
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>
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>
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[asterisk-users] VoIP monitoring tools

[Nitesh Bansal]

Hello all,

The question isn't directly related to Asterisk, but I'm looking for
recommendations
for a monitoring tool to monitor the health of Asterisk instances running
in Production.

Ideally, the tool should be able to generate monitoring traffic (OPTIONS
ping or INVITE),
use the response/no response from Asterisk to store the health of an
Asterisk instance running
somewhere in the DB.

Thanks,
Nitesh Bansal
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[asterisk-users] Propagating the hangup cause between two channels

[Nitesh Bansal]

Hello,

I'm using Asterisk 11 and chan_sip.

Here is the use case:
A dials into Asterisk, Asterisk connects it to B  using Dial command
Under a particular condition, we use AMI to hangup channel A with a
specific *HangUp* Cause.
Asterisk hangs up the channel B, but hangup cause isn't passed from channel
A to B.

Is there anyway in Asterisk for me to propagate the hangup cause between
these two channels?


Thanks,
Nitesh
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Re: [asterisk-users] Dial command for SIP driver with To-header config

[Nitesh Bansal]

Thanks Matt, I adjusted my code to trim the URI scheme.

Regards,
Nitesh

On Tue, Apr 26, 2016 at 2:45 PM, Matthew Jordan <mjor...@digium.com> wrote:

>
> On Fri, Apr 22, 2016 at 11:04 AM, Nitesh Bansal <nitesh.ban...@gmail.com>
> wrote:
>
>> Hello,
>>
>> I'm using the following Dial command syntax:
>> Dial*(SIP/peer/exten!sip:x...@xyz.com <sip%3a...@xyz.com>*), the SIP URI
>> after the '!' mark should be set as To-URI in outgoing INVITE
>> from Asterisk.
>> It works, but problem is that To-URI formatting is a bit messed up,
>> It looks as follows:
>> *sip:sip:x...@xyz.com <sip%3asip%3a...@xyz.com>*, it seems that Asterisk
>> added an extra '*sip:'* in the
>> To-header and it breaks.
>>
>> I'm using Asterisk 13.
>> I'm wondering if this behaviour is intended or a potential bug?
>>
>>
> I would think that it isn't a bug. If you look at the documentation of
> that dial string option for the chan_sip channel driver in sip.conf.sample,
> you can see that the URI scheme is left off:
>
>   54 ; All of these dial strings specify the SIP request URI.
>   55 ; In addition, you can specify a specific To: header by adding an
>   56 ; exclamation mark after the dial string, like
>   57 ;
>   58 ; SIP/sales@mysipproxy!sa...@edvina.net
>
> While it might be nice if it didn't always use a scheme of 'sip', that'd
> probably be categorized as an improvement to this option.
>
> --
> Matthew Jordan
> Digium, Inc. | Director of Technology
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
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[asterisk-users] Dial command for SIP driver with To-header config

[Nitesh Bansal]

Hello,

I'm using the following Dial command syntax:
Dial*(SIP/peer/exten!sip:x...@xyz.com *), the SIP URI
after the '!' mark should be set as To-URI in outgoing INVITE
from Asterisk.
It works, but problem is that To-URI formatting is a bit messed up,
It looks as follows:
*sip:sip:x...@xyz.com *, it seems that Asterisk
added an extra '*sip:'* in the
To-header and it breaks.

I'm using Asterisk 13.
I'm wondering if this behaviour is intended or a potential bug?

Thanks,
Nitesh
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Re: [asterisk-users] [asterisk-dev] Configuring Request URI with outbound proxyu

[Nitesh Bansal]

Hi Olle,

Redirecting the question to users mailing list.
Could you point out how can I *dynamically* pass both the SIP peer and
request-URI in the dial command.
I want be able to use same SIP peer to route to different SIP end points.
I'm currently doing this 'Dial, SIP/peer/exten)', but this results in
Reqeust-URI looking like this sip:exten@ipaddress_of_peer,
whereas I want to be able to somehow pass the SIP request URI to the Dial
command, I tried passing it as part of Route header
in the Dial command and use Kamailio to do loose_route(), but I suppose
this isn't the best solution.

Many thanks,
Nitesh

On Wed, Apr 13, 2016 at 10:17 PM, Olle E. Johansson <o...@edvina.net> wrote:

>
> On 13 Apr 2016, at 22:05, Nitesh Bansal <nitesh.ban...@gmail.com> wrote:
>
> Hello,
>
> I want to use Asterisk to use Kamailio as an outbound proxy for routing
> calls to remote SIP end points, one option could be to use a default peer,
> but in my case, my outbound proxy can change
> based on the remote end point, so this option doesn't work.
> And another problem is that I don't know how to configure Asterisk to
> prepare the Request-URI
> based on the remote end point and not based on the outbound proxy address?
>
> What is the best way to do it?
>
>
> First, you are asking the wrong mailing list. This is not second-level
> support - this is for development and code questions.
>
> If you are using chan_sip, there is a setting of outbound proxy per peer.
> There are settings
> for domain - both in r-uri (host) and from URI domain.
>
> If you set host=example.com and outbound proxy to example.net the SIP
> request will
> have example.com in the R-uri and send the request to example.net
>
> Good luck working with Kamailio!
>
> /Olle
>
>
>
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Re: [asterisk-users] Using Asterisk to route call via an outbound proxy

[Nitesh Bansal]

I'm also using ARI to dynamically select the Asterisk peer and remote end
point, concern is how to use the same peer configuration
with different end points and have asterisk populate the request-uri
correctly?

Nitesh

On Wed, Apr 13, 2016 at 10:11 PM, Nitesh Bansal <nitesh.ban...@gmail.com>
wrote:

> I'm using chan_sip, I experimented with adding a 'Route' header in the
> originate command and used the Dial command like 'SIP/peer/exten', but
> problem
> is that Request-URI isn't populated correctly.
> I'm using Asterisk 13.
>
> Thanks,
> Nitesh
>
> On Wed, Apr 13, 2016 at 10:09 PM, Joshua Colp <jc...@digium.com> wrote:
>
>> Nitesh Bansal wrote:
>>
>>> Hello,
>>>
>>> I want to use Asterisk to use Kamailio as an outbound proxy for routing
>>> calls to remote SIP end points, one option could be to use a default
>>> peer, but in my case, my outbound proxy can change
>>> based on the remote end point, so this option doesn't work.
>>> And another problem is that I don't know how to configure Asterisk to
>>> prepare the Request-URI
>>> based on the remote end point and not based on the outbound proxy
>>> address?
>>>
>>> What is the best way to do it?
>>>
>>
>> You'll have to be specific with the channel driver in use. Speaking from
>> chan_pjsip it does not have a mechanism to set the outbound proxy on a
>> per-call basis, it's strictly controlled by the endpoint. You'd need
>> multiple or construct endpoints dynamically (for example using the ARI push
>> configuration). As for not rewriting the request URI you need to use loose
>> routing by specifying ;lr in the outbound proxy URI.
>>
>> Example:
>>
>> sip:example.com;lr
>>
>> If used in a configuration file:
>>
>> sip:example.com\;lr
>>
>> The '\' is so the configuration parser does not treat it as a comment.
>>
>> Cheers,
>>
>> --
>> Joshua Colp
>> Digium, Inc. | Senior Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & www.asterisk.org
>>
>> --
>> _
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>>   http://www.asterisk.org/hello
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] Using Asterisk to route call via an outbound proxy

[Nitesh Bansal]

I'm using chan_sip, I experimented with adding a 'Route' header in the
originate command and used the Dial command like 'SIP/peer/exten', but
problem
is that Request-URI isn't populated correctly.
I'm using Asterisk 13.

Thanks,
Nitesh

On Wed, Apr 13, 2016 at 10:09 PM, Joshua Colp <jc...@digium.com> wrote:

> Nitesh Bansal wrote:
>
>> Hello,
>>
>> I want to use Asterisk to use Kamailio as an outbound proxy for routing
>> calls to remote SIP end points, one option could be to use a default
>> peer, but in my case, my outbound proxy can change
>> based on the remote end point, so this option doesn't work.
>> And another problem is that I don't know how to configure Asterisk to
>> prepare the Request-URI
>> based on the remote end point and not based on the outbound proxy address?
>>
>> What is the best way to do it?
>>
>
> You'll have to be specific with the channel driver in use. Speaking from
> chan_pjsip it does not have a mechanism to set the outbound proxy on a
> per-call basis, it's strictly controlled by the endpoint. You'd need
> multiple or construct endpoints dynamically (for example using the ARI push
> configuration). As for not rewriting the request URI you need to use loose
> routing by specifying ;lr in the outbound proxy URI.
>
> Example:
>
> sip:example.com;lr
>
> If used in a configuration file:
>
> sip:example.com\;lr
>
> The '\' is so the configuration parser does not treat it as a comment.
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
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[asterisk-users] Using Asterisk to route call via an outbound proxy

[Nitesh Bansal]

Hello,

I want to use Asterisk to use Kamailio as an outbound proxy for routing
calls to remote SIP end points, one option could be to use a default peer,
but in my case, my outbound proxy can change
based on the remote end point, so this option doesn't work.
And another problem is that I don't know how to configure Asterisk to
prepare the Request-URI
based on the remote end point and not based on the outbound proxy address?

What is the best way to do it?

Thanks,
Nitesh
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