Replacing Asterisk with Kamailio - sr-users

anfecora

13 Sep 13 Sep

12:07 p.m.

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(a)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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Gholamreza Sabery

2:11 p.m.

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(a)fastway.com.br> wrote:

...

We won't need transcoding. Is b2b b2bua? Em 13 de set de 2016 13:07, "anfecora" <anfecora(a)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(a)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(a)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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Valter Nogueira

14 Sep 14 Sep

10:59 p.m.

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(a)gmail.com>om>:

...

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(a)gmail.com>om>:

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(a)fastway.com.br> wrote:

We won't need transcoding. Is b2b b2bua? Em 13 de set de 2016 13:07, "anfecora" <anfecora(a)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(a)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(a)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(a)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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Valter Nogueira

11:24 a.m.

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(a)gmail.com>om>:

...

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(a)fastway.com.br>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(a)gmail.com>om>:

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(a)gmail.com>om>:

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(a)fastway.com.br > wrote: > We won't need transcoding. > > Is b2b b2bua? > > Em 13 de set de 2016 13:07, "anfecora" <anfecora(a)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(a)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(a)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(a)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(a)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(a)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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Valter Nogueira

11:31 a.m.

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(a)gmail.com>om>:

...

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(a)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(a)gmail.com>om>:

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(a)gmail.com>om>:

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(a)fastway.com.br > wrote: > We won't need transcoding. > > Is b2b b2bua? > > Em 13 de set de 2016 13:07, "anfecora" <anfecora(a)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(a)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(a)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(a)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(a)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(a)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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Valter Nogueira

17 Sep 17 Sep

7:46 p.m.

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(a)gmail.com>om>:

...

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(a)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(a)gmail.com>om>:

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(a)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(a)gmail.com>om>: > 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(a)gmail.com>om>: > >> 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(a)fastway.com.br> wrote: >> >>> We won't need transcoding. >>> >>> Is b2b b2bua? >>> >>> Em 13 de set de 2016 13:07, "anfecora" <anfecora(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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