Re: [SR-Users] Replacing Asterisk with Kamailio

2016-09-17 Thread Valter Nogueira
also try to understand the pseudo variables which will help you pin >>> point each header within the configuration for example. >>> >>> hope that helps, it is not an easy dry cut learning kamailio neither is >>> to understand the rfc3261, but you must understand

Re: [SR-Users] Replacing Asterisk with Kamailio

2016-09-15 Thread Valter Nogueira
ill help you pin point > each header within the configuration for example. > > hope that helps, it is not an easy dry cut learning kamailio neither is to > understand the rfc3261, but you must understand both to master the > technology. > > > > On Wed, Sep 14, 2016 at 10:

Re: [SR-Users] Replacing Asterisk with Kamailio

2016-09-15 Thread Valter Nogueira
must think not about dialplan there but about method handling. And it > need to very good know SIP RFC for understanding what is going on and why. > > I suppose everyone who uses kamailio thought before thant knows SIP. But > it was wrong. > > 2016-09-15 5:59 GMT+03:00 Valter Nogu

Re: [SR-Users] Replacing Asterisk with Kamailio

2016-09-14 Thread Valter Nogueira
purpose you can use example config file it is a very good >> place to start. Also if you want automatic installation and deployment you >> can use this project: >> >> https://github.com/ghrst/Kamailio-HA >> >> >> On Tue, Sep 13, 2016 at 8:57 PM, Valter Nogueira >

Re: [SR-Users] Replacing Asterisk with Kamailio

2016-09-13 Thread Valter Nogueira
io setup in the same server. > > On Sep 13, 2016 8:42 AM, "Valter Nogueira" wrote: > >> I use Asterisk for SIP and Media Proxy. Despite the fact that Asterisk is >> not a SIP Proxy at all. >> >> Customer registers in a SIP account, sends the invite and thr

[SR-Users] Replacing Asterisk with Kamailio

2016-09-13 Thread Valter Nogueira
I use Asterisk for SIP and Media Proxy. Despite the fact that Asterisk is not a SIP Proxy at all. Customer registers in a SIP account, sends the invite and thru de context Asterisk dials out thru a SIP Trunk. Asterisk does the media proxy, since customer can't route directly to the SIP Trunk (alto