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Re: [SR-Users] SBC

Kamailio v3.3.4 Released

planning to release Kamailio v3.3.4

Keith

19 Feb 2013 19 Feb '13

10:52 a.m.

Hi, I am looking to build an SBC/SIP router made up from Kamailio and FreePBX. Calls will then need to be passed to multiple Asterisk media gateways. Can anyone point me to some documentation or help on how to do this? -- Keith

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Radek Wal

20 Feb 20 Feb

4:24 p.m.

New subject: SBC

SEMS als SBC application. . Am 20.02.2013 um 09:38 schrieb Keith <keithhubner(a)gmail.com>om>:

Hi, I am looking to build an SBC/SIP router made up from Kamailio and FreePBX. Calls will then need to be passed to multiple Asterisk media gateways. Can anyone point me to some documentation or help on how to do this? -- Keith _______________________________________________ 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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Andreas Granig

8:12 p.m.

New subject: SBC

Hi, On 02/20/2013 04:24 PM, Radek Wal wrote:

SEMS als SBC application.

You might want to check http://www.sipwise.com/news/technical/byov-system-spce-as-sbc/ which uses kamailio and sems to do exactly that. Andreas

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Keith

25 Feb 25 Feb

12:16 p.m.

New subject: SBC

Thanks, any reason I could not just use FreeSwitch as an SBC and then load calls between multiple Asterisk servers? Thanks, Keith On Wed, Feb 20, 2013 at 7:12 PM, Andreas Granig <agranig(a)sipwise.com> wrote:

Hi, On 02/20/2013 04:24 PM, Radek Wal wrote:

SEMS als SBC application.

You might want to check http://www.sipwise.com/news/** technical/byov-system-spce-as-**sbc/<http://www.sipwise.com/news/technic… uses kamailio and sems to do exactly that. Andreas

-- Keith

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Alex Balashov

26 Feb 26 Feb

9:41 a.m.

New subject: SBC

There is no reason, other than the "soft" question of FreeSwitch's suitability to the SBC role, as the term has been defined by the industry, and the inertial forces of custom and habit. Keith <keithhubner(a)gmail.com> wrote:

Thanks, any reason I could not just use FreeSwitch as an SBC and then load calls between multiple Asterisk servers? Thanks, Keith On Wed, Feb 20, 2013 at 7:12 PM, Andreas Granig <agranig(a)sipwise.com> wrote:

Hi, On 02/20/2013 04:24 PM, Radek Wal wrote:

SEMS als SBC application.

You might want to check http://www.sipwise.com/news/**

technical/byov-system-spce-as-**sbc/<http://www.sipwise.com/news/technic… uses kamailio and sems to do exactly that.

Andreas

-- Keith ------------------------------------------------------------------------ _______________________________________________ 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

-- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Keith

9:45 a.m.

New subject: SBC

So in my setup where FreeSwitch and Kamailio sit on the same physical server: Kamailio - Routes traffic to asterisk boxes Freeswitch - SBC and RTP traffic B2BUA Question is, can I load balance my voice traffic to the asterisk servers?

From what I have read you can only do this with a percentage rather than

based on the amount of traffic currently on each server (true load balancing). Thanks, Keith On Tue, Feb 26, 2013 at 8:41 AM, Alex Balashov <abalashov(a)evaristesys.com>wrote;wrote:

There is no reason, other than the "soft" question of FreeSwitch's suitability to the SBC role, as the term has been defined by the industry, and the inertial forces of custom and habit. Keith <keithhubner(a)gmail.com> wrote:

Thanks, any reason I could not just use FreeSwitch as an SBC and then load calls between multiple Asterisk servers? Thanks, Keith On Wed, Feb 20, 2013 at 7:12 PM, Andreas Granig <agranig(a)sipwise.com>wrote;wrote:

Hi, On 02/20/2013 04:24 PM, Radek Wal wrote:

SEMS als SBC application.

You might want to check http://www.sipwise.com/news/** technical/byov-system-spce-as-**sbc/<http://www.sipwise.com/news/technic… uses kamailio and sems to do exactly that. Andreas

-- Keith ------------------------------ 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

-- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

-- Keith

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davy van de moere

20 Feb 20 Feb

4:27 p.m.

New subject: SBC

Stampeding on an open door here, and not wanting to start a pun war ;) http://www.amazon.com/Building-Telephony-Systems-OpenSIPS-1-6/dp/1849510741… Offcourse, it's not *AS* good as Kamailio, but if your into the book thing, it might help you to get your head around how the thing actually works. The question you ask, is actually a close to default setting of Kamailio, so I think you need to or: 1/ get your head around routing sip packets, which at starters is mindblowing, so give yourself some time. 2/ pay someone to set a kamailio up for you which does what you want. (I think almost anyone on the mailinglist here, can do that in a matter of hours) Enjoy! On 19 Feb 2013, at 10:52, Keith wrote:

Hi, I am looking to build an SBC/SIP router made up from Kamailio and FreePBX. Calls will then need to be passed to multiple Asterisk media gateways. Can anyone point me to some documentation or help on how to do this? -- Keith _______________________________________________ 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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andrzej.ciupek＠asterisk.edu.pl

5:01 p.m.

New subject: SBC

Hello Or start with this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb later You can change it to use dispatcher module to have load balance between couple od Asterisk GW. But try to change configuration to use WITH_PSTN, not WITH_ASTERISK, that won't forward registration to Asterisk, only INVITE's, in PSTN Route implement dispatcher module. Greetings Andrzej Cytowanie davy van de moere <davy.van.de.moere(a)gmail.com>om>:

Stampeding on an open door here, and not wanting to start a pun war ;) http://www.amazon.com/Building-Telephony-Systems-OpenSIPS-1-6/dp/1849510741… Offcourse, it's not *AS* good as Kamailio, but if your into the book thing, it might help you to get your head around how the thing actually works. The question you ask, is actually a close to default setting of Kamailio, so I think you need to or: 1/ get your head around routing sip packets, which at starters is mindblowing, so give yourself some time. 2/ pay someone to set a kamailio up for you which does what you want. (I think almost anyone on the mailinglist here, can do that in a matter of hours) Enjoy! On 19 Feb 2013, at 10:52, Keith wrote:

Hi, I am looking to build an SBC/SIP router made up from Kamailio and FreePBX. Calls will then need to be passed to multiple Asterisk media gateways. Can anyone point me to some documentation or help on how to do this? -- Keith _______________________________________________ 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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participants (6)

Alex Balashov
Andreas Granig
andrzej.ciupek＠asterisk.edu.pl
davy van de moere
Keith
Radek Wal