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 (altough it has a valida address, it don't have a public route allowed to it).
I need limit customer concurrent calls, mangle some dial-in/dial-out numbers, keep track of ongoing call, control SIP dialog, retransmit correct hang-up causes and do media proxy (no transconding at all)
After reading about Kamailio and Opensips, and due to the Kamailio Admin Book, I decided to go with Kamailio.
Well, I understand that I have to use some kamailio modules, like auth, dialplan, rtpproxy and db_mysql.
What make me stuck is how does everything fit together in kamailio.cfg and how do I get ongoing calls and CDR's?
Can anyone point me a direction?
Thanks
Valter i wouldnt take fully asterisk from the picture you can use it to handle transcoding for example and still a b2b support.
Perhaps you can look for asterisk kamailio setup in the same server.
On Sep 13, 2016 8:42 AM, "Valter Nogueira" valter@fastway.com.br 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 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 (altough it has a valida address, it don't have a public route allowed to it).
I need limit customer concurrent calls, mangle some dial-in/dial-out numbers, keep track of ongoing call, control SIP dialog, retransmit correct hang-up causes and do media proxy (no transconding at all)
After reading about Kamailio and Opensips, and due to the Kamailio Admin Book, I decided to go with Kamailio.
Well, I understand that I have to use some kamailio modules, like auth, dialplan, rtpproxy and db_mysql.
What make me stuck is how does everything fit together in kamailio.cfg and how do I get ongoing calls and CDR's?
Can anyone point me a direction?
Thanks
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
We won't need transcoding.
Is b2b b2bua?
Em 13 de set de 2016 13:07, "anfecora" anfecora@gmail.com escreveu:
Valter i wouldnt take fully asterisk from the picture you can use it to handle transcoding for example and still a b2b support.
Perhaps you can look for asterisk kamailio setup in the same server.
On Sep 13, 2016 8:42 AM, "Valter Nogueira" valter@fastway.com.br 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 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 (altough it has a valida address, it don't have a public route allowed to it).
I need limit customer concurrent calls, mangle some dial-in/dial-out numbers, keep track of ongoing call, control SIP dialog, retransmit correct hang-up causes and do media proxy (no transconding at all)
After reading about Kamailio and Opensips, and due to the Kamailio Admin Book, I decided to go with Kamailio.
Well, I understand that I have to use some kamailio modules, like auth, dialplan, rtpproxy and db_mysql.
What make me stuck is how does everything fit together in kamailio.cfg and how do I get ongoing calls and CDR's?
Can anyone point me a direction?
Thanks
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
For testing 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 valter@fastway.com.br wrote:
We won't need transcoding.
Is b2b b2bua?
Em 13 de set de 2016 13:07, "anfecora" anfecora@gmail.com escreveu:
Valter i wouldnt take fully asterisk from the picture you can use it to handle transcoding for example and still a b2b support.
Perhaps you can look for asterisk kamailio setup in the same server.
On Sep 13, 2016 8:42 AM, "Valter Nogueira" valter@fastway.com.br 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 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 (altough it has a valida address, it don't have a public route allowed to it).
I need limit customer concurrent calls, mangle some dial-in/dial-out numbers, keep track of ongoing call, control SIP dialog, retransmit correct hang-up causes and do media proxy (no transconding at all)
After reading about Kamailio and Opensips, and due to the Kamailio Admin Book, I decided to go with Kamailio.
Well, I understand that I have to use some kamailio modules, like auth, dialplan, rtpproxy and db_mysql.
What make me stuck is how does everything fit together in kamailio.cfg and how do I get ongoing calls and CDR's?
Can anyone point me a direction?
Thanks
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
it is many-many examples of kamialio.cfg at the internet that describes same logic with different staff (like kamailio as registrar and also as kamailio as just proxy)
I suppose you just dont fully understood logic of how kamailo working.
Just goole first. I aslo had same question some time ago. google helped me to understand all it. really. Just trying to help
Read this
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb http://lextertech.blogspot.ru/2015/01/asterisk-v117-realtime-integration-wit... https://www.kamailio.org/dokuwiki/doku.php/asterisk:realtime-integration
and this (dont see that it is old.Logis is the same)
https://www.kamailio.org/w/2010/11/asterisk-1-6-and-kamailio-3-1-realtime-in...
All this just one of the many variants how you can to integrate it. Good Luck. I suppose you will know many new cool things when open kamailio for yourself.
2016-09-13 21:11 GMT+03:00 Gholamreza Sabery gr.sabery@gmail.com:
For testing 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 valter@fastway.com.br wrote:
We won't need transcoding.
Is b2b b2bua?
Em 13 de set de 2016 13:07, "anfecora" anfecora@gmail.com escreveu:
Valter i wouldnt take fully asterisk from the picture you can use it to handle transcoding for example and still a b2b support.
Perhaps you can look for asterisk kamailio setup in the same server.
On Sep 13, 2016 8:42 AM, "Valter Nogueira" valter@fastway.com.br 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 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 (altough it has a valida address, it don't have a public route allowed to it).
I need limit customer concurrent calls, mangle some dial-in/dial-out numbers, keep track of ongoing call, control SIP dialog, retransmit correct hang-up causes and do media proxy (no transconding at all)
After reading about Kamailio and Opensips, and due to the Kamailio Admin Book, I decided to go with Kamailio.
Well, I understand that I have to use some kamailio modules, like auth, dialplan, rtpproxy and db_mysql.
What make me stuck is how does everything fit together in kamailio.cfg and how do I get ongoing calls and CDR's?
Can anyone point me a direction?
Thanks
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Yes, you are absolutely right: I don't understand (yet) how Kamailio works!
I prefer installing it from sources.
What I get until now, is that kamailio.cfg is more a program than a configuration file at all.
I really appreciate the links and I will try to understand them.
Thank you
Valter
2016-09-13 16:11 GMT-03:00 Yuriy Gorlichenko ovoshlook@gmail.com:
it is many-many examples of kamialio.cfg at the internet that describes same logic with different staff (like kamailio as registrar and also as kamailio as just proxy)
I suppose you just dont fully understood logic of how kamailo working.
Just goole first. I aslo had same question some time ago. google helped me to understand all it. really. Just trying to help
Read this
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x- asterisk-11.3.0-astdb http://lextertech.blogspot.ru/2015/01/asterisk-v117- realtime-integration-with.html https://www.kamailio.org/dokuwiki/doku.php/asterisk:realtime-integration
and this (dont see that it is old.Logis is the same)
https://www.kamailio.org/w/2010/11/asterisk-1-6-and-kamailio-3-1-realtime- integration-tutorial/
All this just one of the many variants how you can to integrate it. Good Luck. I suppose you will know many new cool things when open kamailio for yourself.
2016-09-13 21:11 GMT+03:00 Gholamreza Sabery gr.sabery@gmail.com:
For testing 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 valter@fastway.com.br wrote:
We won't need transcoding.
Is b2b b2bua?
Em 13 de set de 2016 13:07, "anfecora" anfecora@gmail.com escreveu:
Valter i wouldnt take fully asterisk from the picture you can use it to handle transcoding for example and still a b2b support.
Perhaps you can look for asterisk kamailio setup in the same server.
On Sep 13, 2016 8:42 AM, "Valter Nogueira" valter@fastway.com.br 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 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 (altough it has a valida address, it don't have a public route allowed to it).
I need limit customer concurrent calls, mangle some dial-in/dial-out numbers, keep track of ongoing call, control SIP dialog, retransmit correct hang-up causes and do media proxy (no transconding at all)
After reading about Kamailio and Opensips, and due to the Kamailio Admin Book, I decided to go with Kamailio.
Well, I understand that I have to use some kamailio modules, like auth, dialplan, rtpproxy and db_mysql.
What make me stuck is how does everything fit together in kamailio.cfg and how do I get ongoing calls and CDR's?
Can anyone point me a direction?
Thanks
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Yes. Thats correct Kamailio.cfg is a script. It is a sublnguage. You must think as programmer for using it
Input data is sip method. Thist script using it for handling making changes if it need and make him to know where it must be proxyed That is a main idea.
Actually in asteirks for example it is a same btut asteirsk works at the extensions.conf/ael/lua with invite method and Taking Leg A and creates Leg B
kamailio is different. It is just proxies same method through itself. and working with every method like invite and his replies, and other methods.
You 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 Nogueira valter@fastway.com.br:
Yes, you are absolutely right: I don't understand (yet) how Kamailio works!
I prefer installing it from sources.
What I get until now, is that kamailio.cfg is more a program than a configuration file at all.
I really appreciate the links and I will try to understand them.
Thank you
Valter
2016-09-13 16:11 GMT-03:00 Yuriy Gorlichenko ovoshlook@gmail.com:
it is many-many examples of kamialio.cfg at the internet that describes same logic with different staff (like kamailio as registrar and also as kamailio as just proxy)
I suppose you just dont fully understood logic of how kamailo working.
Just goole first. I aslo had same question some time ago. google helped me to understand all it. really. Just trying to help
Read this
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asteri sk-11.3.0-astdb http://lextertech.blogspot.ru/2015/01/asterisk-v117-realtime -integration-with.html https://www.kamailio.org/dokuwiki/doku.php/asterisk:realtime-integration
and this (dont see that it is old.Logis is the same)
https://www.kamailio.org/w/2010/11/asterisk-1-6-and-kamailio -3-1-realtime-integration-tutorial/
All this just one of the many variants how you can to integrate it. Good Luck. I suppose you will know many new cool things when open kamailio for yourself.
2016-09-13 21:11 GMT+03:00 Gholamreza Sabery gr.sabery@gmail.com:
For testing 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 valter@fastway.com.br wrote:
We won't need transcoding.
Is b2b b2bua?
Em 13 de set de 2016 13:07, "anfecora" anfecora@gmail.com escreveu:
Valter i wouldnt take fully asterisk from the picture you can use it to handle transcoding for example and still a b2b support.
Perhaps you can look for asterisk kamailio setup in the same server.
On Sep 13, 2016 8:42 AM, "Valter Nogueira" valter@fastway.com.br 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 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 (altough it has a valida address, it don't have a public route allowed to it).
I need limit customer concurrent calls, mangle some dial-in/dial-out numbers, keep track of ongoing call, control SIP dialog, retransmit correct hang-up causes and do media proxy (no transconding at all)
After reading about Kamailio and Opensips, and due to the Kamailio Admin Book, I decided to go with Kamailio.
Well, I understand that I have to use some kamailio modules, like auth, dialplan, rtpproxy and db_mysql.
What make me stuck is how does everything fit together in kamailio.cfg and how do I get ongoing calls and CDR's?
Can anyone point me a direction?
Thanks
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
When a call cames from my customer, it will be autheticated by auth module.
After authetication, I need to proxy it to the sip provider, but it expects my credentials, not my customer's one.
It not seems to me, correct if I am wrong, a proxy operation, is it?
Since I am a programmer, I expected to have little problem to undertand kamailio.cfg, but when I read some code, I wonder how could someone think in this sip sequence and how could do I know that I am doing right?
Atenciosamente,
2016-09-15 3:04 GMT-03:00 Yuriy Gorlichenko ovoshlook@gmail.com:
Yes. Thats correct Kamailio.cfg is a script. It is a sublnguage. You must think as programmer for using it
Input data is sip method. Thist script using it for handling making changes if it need and make him to know where it must be proxyed That is a main idea.
Actually in asteirks for example it is a same btut asteirsk works at the extensions.conf/ael/lua with invite method and Taking Leg A and creates Leg B
kamailio is different. It is just proxies same method through itself. and working with every method like invite and his replies, and other methods.
You 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 Nogueira valter@fastway.com.br:
Yes, you are absolutely right: I don't understand (yet) how Kamailio works!
I prefer installing it from sources.
What I get until now, is that kamailio.cfg is more a program than a configuration file at all.
I really appreciate the links and I will try to understand them.
Thank you
Valter
2016-09-13 16:11 GMT-03:00 Yuriy Gorlichenko ovoshlook@gmail.com:
it is many-many examples of kamialio.cfg at the internet that describes same logic with different staff (like kamailio as registrar and also as kamailio as just proxy)
I suppose you just dont fully understood logic of how kamailo working.
Just goole first. I aslo had same question some time ago. google helped me to understand all it. really. Just trying to help
Read this
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asteri sk-11.3.0-astdb http://lextertech.blogspot.ru/2015/01/asterisk-v117-realtime -integration-with.html https://www.kamailio.org/dokuwiki/doku.php/asterisk:realtime-integration
and this (dont see that it is old.Logis is the same)
https://www.kamailio.org/w/2010/11/asterisk-1-6-and-kamailio -3-1-realtime-integration-tutorial/
All this just one of the many variants how you can to integrate it. Good Luck. I suppose you will know many new cool things when open kamailio for yourself.
2016-09-13 21:11 GMT+03:00 Gholamreza Sabery gr.sabery@gmail.com:
For testing 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 <valter@fastway.com.br
wrote:
We won't need transcoding.
Is b2b b2bua?
Em 13 de set de 2016 13:07, "anfecora" anfecora@gmail.com escreveu:
Valter i wouldnt take fully asterisk from the picture you can use it to handle transcoding for example and still a b2b support.
Perhaps you can look for asterisk kamailio setup in the same server.
On Sep 13, 2016 8:42 AM, "Valter Nogueira" valter@fastway.com.br 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 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 (altough it has a > valida address, it don't have a public route allowed to it). > > I need limit customer concurrent calls, mangle some dial-in/dial-out > numbers, keep track of ongoing call, control SIP dialog, retransmit correct > hang-up causes and do media proxy (no transconding at all) > > After reading about Kamailio and Opensips, and due to the Kamailio > Admin Book, I decided to go with Kamailio. > > Well, I understand that I have to use some kamailio modules, like > auth, dialplan, rtpproxy and db_mysql. > > What make me stuck is how does everything fit together in > kamailio.cfg and how do I get ongoing calls and CDR's? > > Can anyone point me a direction? > > Thanks > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Valter, you can buy the the draft of Kamailio Admin Book. It will really help you understand Kamilio how it works , how it porcess calls and what each part of the cfg means.
yes it is true that you need to know about sip and how request correlate with responses etc because it does correlate the way it is handled.
kamailio.cfg is a very simple configuration file, you define the modules that you want to use and then how to process the calls, it is not as explicit as asterisk because asterisk is a different tool, kamailio is a full proxy/sip router while asterisk is only a registrar.
you can also try to download/ buy the book Building Telephony Systems with OpenSER Flavio E. Goncalves April 2008
which is a lso very explicit on how is the cfg divided.
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 both to master the technology.
On Wed, Sep 14, 2016 at 10:59 PM, Valter Nogueira valter@fastway.com.br wrote:
Yes, you are absolutely right: I don't understand (yet) how Kamailio works!
I prefer installing it from sources.
What I get until now, is that kamailio.cfg is more a program than a configuration file at all.
I really appreciate the links and I will try to understand them.
Thank you
Valter
2016-09-13 16:11 GMT-03:00 Yuriy Gorlichenko ovoshlook@gmail.com:
it is many-many examples of kamialio.cfg at the internet that describes same logic with different staff (like kamailio as registrar and also as kamailio as just proxy)
I suppose you just dont fully understood logic of how kamailo working.
Just goole first. I aslo had same question some time ago. google helped me to understand all it. really. Just trying to help
Read this
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asteri sk-11.3.0-astdb http://lextertech.blogspot.ru/2015/01/asterisk-v117-realtime -integration-with.html https://www.kamailio.org/dokuwiki/doku.php/asterisk:realtime-integration
and this (dont see that it is old.Logis is the same)
https://www.kamailio.org/w/2010/11/asterisk-1-6-and-kamailio -3-1-realtime-integration-tutorial/
All this just one of the many variants how you can to integrate it. Good Luck. I suppose you will know many new cool things when open kamailio for yourself.
2016-09-13 21:11 GMT+03:00 Gholamreza Sabery gr.sabery@gmail.com:
For testing 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 valter@fastway.com.br wrote:
We won't need transcoding.
Is b2b b2bua?
Em 13 de set de 2016 13:07, "anfecora" anfecora@gmail.com escreveu:
Valter i wouldnt take fully asterisk from the picture you can use it to handle transcoding for example and still a b2b support.
Perhaps you can look for asterisk kamailio setup in the same server.
On Sep 13, 2016 8:42 AM, "Valter Nogueira" valter@fastway.com.br 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 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 (altough it has a valida address, it don't have a public route allowed to it).
I need limit customer concurrent calls, mangle some dial-in/dial-out numbers, keep track of ongoing call, control SIP dialog, retransmit correct hang-up causes and do media proxy (no transconding at all)
After reading about Kamailio and Opensips, and due to the Kamailio Admin Book, I decided to go with Kamailio.
Well, I understand that I have to use some kamailio modules, like auth, dialplan, rtpproxy and db_mysql.
What make me stuck is how does everything fit together in kamailio.cfg and how do I get ongoing calls and CDR's?
Can anyone point me a direction?
Thanks
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
I already have both books and I am enrolled in next Flavio's on-line bootcamp.
I have never get the rfc3261 end. But I will try to read it again (may be I have to print it, I am a little old to read such on-line - I guess)
Thank you,
2016-09-15 3:21 GMT-03:00 anfecora anfecora@gmail.com:
Valter, you can buy the the draft of Kamailio Admin Book. It will really help you understand Kamilio how it works , how it porcess calls and what each part of the cfg means.
yes it is true that you need to know about sip and how request correlate with responses etc because it does correlate the way it is handled.
kamailio.cfg is a very simple configuration file, you define the modules that you want to use and then how to process the calls, it is not as explicit as asterisk because asterisk is a different tool, kamailio is a full proxy/sip router while asterisk is only a registrar.
you can also try to download/ buy the book Building Telephony Systems with OpenSER Flavio E. Goncalves April 2008
which is a lso very explicit on how is the cfg divided.
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 both to master the technology.
On Wed, Sep 14, 2016 at 10:59 PM, Valter Nogueira valter@fastway.com.br wrote:
Yes, you are absolutely right: I don't understand (yet) how Kamailio works!
I prefer installing it from sources.
What I get until now, is that kamailio.cfg is more a program than a configuration file at all.
I really appreciate the links and I will try to understand them.
Thank you
Valter
2016-09-13 16:11 GMT-03:00 Yuriy Gorlichenko ovoshlook@gmail.com:
it is many-many examples of kamialio.cfg at the internet that describes same logic with different staff (like kamailio as registrar and also as kamailio as just proxy)
I suppose you just dont fully understood logic of how kamailo working.
Just goole first. I aslo had same question some time ago. google helped me to understand all it. really. Just trying to help
Read this
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asteri sk-11.3.0-astdb http://lextertech.blogspot.ru/2015/01/asterisk-v117-realtime -integration-with.html https://www.kamailio.org/dokuwiki/doku.php/asterisk:realtime-integration
and this (dont see that it is old.Logis is the same)
https://www.kamailio.org/w/2010/11/asterisk-1-6-and-kamailio -3-1-realtime-integration-tutorial/
All this just one of the many variants how you can to integrate it. Good Luck. I suppose you will know many new cool things when open kamailio for yourself.
2016-09-13 21:11 GMT+03:00 Gholamreza Sabery gr.sabery@gmail.com:
For testing 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 <valter@fastway.com.br
wrote:
We won't need transcoding.
Is b2b b2bua?
Em 13 de set de 2016 13:07, "anfecora" anfecora@gmail.com escreveu:
Valter i wouldnt take fully asterisk from the picture you can use it to handle transcoding for example and still a b2b support.
Perhaps you can look for asterisk kamailio setup in the same server.
On Sep 13, 2016 8:42 AM, "Valter Nogueira" valter@fastway.com.br 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 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 (altough it has a > valida address, it don't have a public route allowed to it). > > I need limit customer concurrent calls, mangle some dial-in/dial-out > numbers, keep track of ongoing call, control SIP dialog, retransmit correct > hang-up causes and do media proxy (no transconding at all) > > After reading about Kamailio and Opensips, and due to the Kamailio > Admin Book, I decided to go with Kamailio. > > Well, I understand that I have to use some kamailio modules, like > auth, dialplan, rtpproxy and db_mysql. > > What make me stuck is how does everything fit together in > kamailio.cfg and how do I get ongoing calls and CDR's? > > Can anyone point me a direction? > > Thanks > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
You might want to look into freeSWITCH also, coupled with Kamailio it's great setup.
On Thu, Sep 15, 2016 at 5:31 PM, Valter Nogueira valter@fastway.com.br wrote:
I already have both books and I am enrolled in next Flavio's on-line bootcamp.
I have never get the rfc3261 end. But I will try to read it again (may be I have to print it, I am a little old to read such on-line - I guess)
Thank you,
2016-09-15 3:21 GMT-03:00 anfecora anfecora@gmail.com:
Valter, you can buy the the draft of Kamailio Admin Book. It will really help you understand Kamilio how it works , how it porcess calls and what each part of the cfg means.
yes it is true that you need to know about sip and how request correlate with responses etc because it does correlate the way it is handled.
kamailio.cfg is a very simple configuration file, you define the modules that you want to use and then how to process the calls, it is not as explicit as asterisk because asterisk is a different tool, kamailio is a full proxy/sip router while asterisk is only a registrar.
you can also try to download/ buy the book Building Telephony Systems with OpenSER Flavio E. Goncalves April 2008
which is a lso very explicit on how is the cfg divided.
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 both to master the technology.
On Wed, Sep 14, 2016 at 10:59 PM, Valter Nogueira valter@fastway.com.br wrote:
Yes, you are absolutely right: I don't understand (yet) how Kamailio works!
I prefer installing it from sources.
What I get until now, is that kamailio.cfg is more a program than a configuration file at all.
I really appreciate the links and I will try to understand them.
Thank you
Valter
2016-09-13 16:11 GMT-03:00 Yuriy Gorlichenko ovoshlook@gmail.com:
it is many-many examples of kamialio.cfg at the internet that describes same logic with different staff (like kamailio as registrar and also as kamailio as just proxy)
I suppose you just dont fully understood logic of how kamailo working.
Just goole first. I aslo had same question some time ago. google helped me to understand all it. really. Just trying to help
Read this
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asteri sk-11.3.0-astdb http://lextertech.blogspot.ru/2015/01/asterisk-v117-realtime -integration-with.html https://www.kamailio.org/dokuwiki/doku.php/asterisk:realtime -integration
and this (dont see that it is old.Logis is the same)
https://www.kamailio.org/w/2010/11/asterisk-1-6-and-kamailio -3-1-realtime-integration-tutorial/
All this just one of the many variants how you can to integrate it. Good Luck. I suppose you will know many new cool things when open kamailio for yourself.
2016-09-13 21:11 GMT+03:00 Gholamreza Sabery gr.sabery@gmail.com:
For testing 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 < valter@fastway.com.br> wrote:
We won't need transcoding.
Is b2b b2bua?
Em 13 de set de 2016 13:07, "anfecora" anfecora@gmail.com escreveu:
> Valter i wouldnt take fully asterisk from the picture you can use it > to handle transcoding for example and still a b2b support. > > Perhaps you can look for asterisk kamailio setup in the same server. > > On Sep 13, 2016 8:42 AM, "Valter Nogueira" valter@fastway.com.br > 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 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 (altough it has a >> valida address, it don't have a public route allowed to it). >> >> I need limit customer concurrent calls, mangle some >> dial-in/dial-out numbers, keep track of ongoing call, control SIP dialog, >> retransmit correct hang-up causes and do media proxy (no transconding at >> all) >> >> After reading about Kamailio and Opensips, and due to the Kamailio >> Admin Book, I decided to go with Kamailio. >> >> Well, I understand that I have to use some kamailio modules, like >> auth, dialplan, rtpproxy and db_mysql. >> >> What make me stuck is how does everything fit together in >> kamailio.cfg and how do I get ongoing calls and CDR's? >> >> Can anyone point me a direction? >> >> Thanks >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Yes, our main product is based on Asterisk and we are moving do FS.
But, for this project it is a no go.
Valter
2016-09-16 10:22 GMT-03:00 David Villasmil david.villasmil.work@gmail.com:
You might want to look into freeSWITCH also, coupled with Kamailio it's great setup.
On Thu, Sep 15, 2016 at 5:31 PM, Valter Nogueira valter@fastway.com.br wrote:
I already have both books and I am enrolled in next Flavio's on-line bootcamp.
I have never get the rfc3261 end. But I will try to read it again (may be I have to print it, I am a little old to read such on-line - I guess)
Thank you,
2016-09-15 3:21 GMT-03:00 anfecora anfecora@gmail.com:
Valter, you can buy the the draft of Kamailio Admin Book. It will really help you understand Kamilio how it works , how it porcess calls and what each part of the cfg means.
yes it is true that you need to know about sip and how request correlate with responses etc because it does correlate the way it is handled.
kamailio.cfg is a very simple configuration file, you define the modules that you want to use and then how to process the calls, it is not as explicit as asterisk because asterisk is a different tool, kamailio is a full proxy/sip router while asterisk is only a registrar.
you can also try to download/ buy the book Building Telephony Systems with OpenSER Flavio E. Goncalves April 2008
which is a lso very explicit on how is the cfg divided.
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 both to master the technology.
On Wed, Sep 14, 2016 at 10:59 PM, Valter Nogueira <valter@fastway.com.br
wrote:
Yes, you are absolutely right: I don't understand (yet) how Kamailio works!
I prefer installing it from sources.
What I get until now, is that kamailio.cfg is more a program than a configuration file at all.
I really appreciate the links and I will try to understand them.
Thank you
Valter
2016-09-13 16:11 GMT-03:00 Yuriy Gorlichenko ovoshlook@gmail.com:
it is many-many examples of kamialio.cfg at the internet that describes same logic with different staff (like kamailio as registrar and also as kamailio as just proxy)
I suppose you just dont fully understood logic of how kamailo working.
Just goole first. I aslo had same question some time ago. google helped me to understand all it. really. Just trying to help
Read this
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asteri sk-11.3.0-astdb http://lextertech.blogspot.ru/2015/01/asterisk-v117-realtime -integration-with.html https://www.kamailio.org/dokuwiki/doku.php/asterisk:realtime -integration
and this (dont see that it is old.Logis is the same)
https://www.kamailio.org/w/2010/11/asterisk-1-6-and-kamailio -3-1-realtime-integration-tutorial/
All this just one of the many variants how you can to integrate it. Good Luck. I suppose you will know many new cool things when open kamailio for yourself.
2016-09-13 21:11 GMT+03:00 Gholamreza Sabery gr.sabery@gmail.com:
For testing 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 < valter@fastway.com.br> wrote:
> We won't need transcoding. > > Is b2b b2bua? > > Em 13 de set de 2016 13:07, "anfecora" anfecora@gmail.com > escreveu: > >> Valter i wouldnt take fully asterisk from the picture you can use >> it to handle transcoding for example and still a b2b support. >> >> Perhaps you can look for asterisk kamailio setup in the same server. >> >> On Sep 13, 2016 8:42 AM, "Valter Nogueira" valter@fastway.com.br >> 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 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 (altough it has a >>> valida address, it don't have a public route allowed to it). >>> >>> I need limit customer concurrent calls, mangle some >>> dial-in/dial-out numbers, keep track of ongoing call, control SIP dialog, >>> retransmit correct hang-up causes and do media proxy (no transconding at >>> all) >>> >>> After reading about Kamailio and Opensips, and due to the Kamailio >>> Admin Book, I decided to go with Kamailio. >>> >>> Well, I understand that I have to use some kamailio modules, like >>> auth, dialplan, rtpproxy and db_mysql. >>> >>> What make me stuck is how does everything fit together in >>> kamailio.cfg and how do I get ongoing calls and CDR's? >>> >>> Can anyone point me a direction? >>> >>> Thanks >>> >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >>> list >>> sr-users@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On Tue, Sep 13, 2016 at 12:42:23PM -0300, Valter Nogueira wrote:
I use Asterisk for SIP and Media Proxy. Despite the fact that Asterisk is not a SIP Proxy at all.
[...]
Well, I understand that I have to use some kamailio modules, like auth, dialplan, rtpproxy and db_mysql.
What make me stuck is how does everything fit together in kamailio.cfg and how do I get ongoing calls and CDR's?
Can anyone point me a direction?
I was once at the exact same point you are. And what helped my the most was the Kamailio Advanced Training course. If you don't want/cannot to do that, there is a kamailio-advanced.cfg provided with the sources/packages. That does all you want except dialplan and dialog modules. Start by dissecting that config.
Start small and build up the config. Just about any question you might have is already asked and hopefully answered on this mailing list, so search the archive and ask what you can't find/understand/figure out. There might be some answers in the cookbooks/tutorials on http://www.kamailio.org/wiki/start#cookbooks
BTW I still use asterisk for upstream SIP trunks, it's b2bua features can be really usefull when talking to the outside world. So again start small and have kamailio talk to your existing asterisk machines (e.g. via the dispatcher module) to reach the outside. Verify your kamailio CDRs (extraced from the accounting module) to asterisk.
-
anfecora -
Daniel Tryba -
David Villasmil -
Gholamreza Sabery -
Valter Nogueira -
Yuriy Gorlichenko