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

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

Re: [asterisk-users] VoIP monitoring tools

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,

Re: [asterisk-users] VoIP monitoring tools

ur 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:

[asterisk-users] VoIP monitoring tools

/no response from Asterisk to store the health of an Asterisk instance running somewhere in the DB. Thanks, Nitesh Bansal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community

[asterisk-users] Propagating the hangup cause between two channels

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

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

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, >> >

[asterisk-users] Dial command for SIP driver with To-header config

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:

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

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 > ca

Re: [asterisk-users] Using Asterisk to route call via an outbound proxy

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.

Re: [asterisk-users] Using Asterisk to route call via an outbound proxy

;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 >&g

[asterisk-users] Using Asterisk to route call via an outbound proxy

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