Thanks, I went for the permission + dispatcher, and so far it looks good.

One last thing.... I need the kamailio server to explicitly register to a provider. Googling the archive, I could only find an old reference to use sipsak. Does that mean that sipsak (or any other sip crafting tool like sipsak) should send a register request to Kamailio, which in turn will forward it to the voip provider ?

Here is the message: https://lists.kamailio.org/pipermail/sr-users/2004-November/041304.html

Rgds J

On Tue, Jun 13, 2017 at 3:01 AM, Daniel-Constantin Mierla <miconda@gmail.com

wrote:

Hello,

I typically blend several modules to achieve your routing needs:

• permissions module with address table to detect if calls are coming

from providers or asterisk farm

• based on result use load balancer to send out to asterisk farm

• or use pdt (if each provider has one gateway) or mtree+dispatcher

(mtree to translate from a prefix to a dispatcher set id of the provider with many gateways) to send out to providers

If least cost routing is needed towards providers, then you can decide to use one of next modules:

• lcr

• carrierroute

• drouting

Cheers, Daniel

On 12.06.17 21:21, Jean Cérien wrote:

Hello

I am looking at the following configuration:

1/ voip providers, with different external sip servers, will send calls to kamailio, that will dispatch them to asterisk servers, with load balancing & failover. The IP of the asterisk servers are known and fixed.

2/ the asterisk servers will also place calls, going through kamailio, that will dispatch them to the voip provider sip servers, based on R-URI / Prefix.

I've achieved #1 with dispatcher module, I've seen carrierroute module that can achieve #2, but I dont really grasp how to put the two together.

I have the feeling that #1 can also be done with carrierroute, and that I only need to check if the 'from' domain is one my asterisk, and chose a different routing table, but I dont see how to do that

is that a valid approch ? where should I look to do that ?

thanks

J.

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Thanks

I have the registration working, but when my kamailio receives an INVITE, I am a bit lost regarding what to do. in my route(AUTH), I am trying to check uac_reg_status, but no success.

my db: kamailio:/etc/kamailio# kamcmd uac.reg_dump { l_uuid: 123 l_username: passgwada l_domain: 192.168.2.228 r_username: passgwada r_domain: 192.168.2.230 realm: xivo auth_username: passgwada auth_password: 1234 auth_proxy: sip:192.168.2.230 expires: 60 flags: 2 diff_expires: -1497475389 timer_expires: 0 reg_init: 1497475386 reg_delay: 5 }

and the invite arriving INVITE sip:1100@192.168.2.228 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.230:5060;branch=z9hG4bK17eeae32;rport Max-Forwards: 70 From: "0512345678" sip:0512345678@192.168.2.230;tag=as0541bcd3 To: sip:1100@192.168.2.228 Contact: sip:0512345678@192.168.2.230:5060 Call-ID: 0bf6e5ac4fe10a0e3478a6dc1c70d40d@192.168.2.230:5060 CSeq: 102 INVITE User-Agent: XiVO PBX Date: Wed, 14 Jun 2017 21:09:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 300 ......

Any help will be greatly appreciated

J.

On Wed, Jun 14, 2017 at 12:45 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:

Sipsak or other tool pointing via Contact header to Kamailio is still an option, but you can use directly uac module -- you have to set the details for remote registrations inside uac_reg database table and configure uac module, some details also at:

• https://kamailio.org/docs/modules/5.0.x/modules/uac.html#idp38102932

Cheers, Daniel

On 14.06.17 18:17, Jean Cérien wrote:

Thanks, I went for the permission + dispatcher, and so far it looks good.

One last thing.... I need the kamailio server to explicitly register to a provider. Googling the archive, I could only find an old reference to use sipsak. Does that mean that sipsak (or any other sip crafting tool like sipsak) should send a register request to Kamailio, which in turn will forward it to the voip provider ?

Here is the message: https://lists.kamailio.org/pipermail/sr- users/2004-November/041304.html

Rgds J

On Tue, Jun 13, 2017 at 3:01 AM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:

...

Hello,

I typically blend several modules to achieve your routing needs:

• permissions module with address table to detect if calls are coming

from providers or asterisk farm

• based on result use load balancer to send out to asterisk farm

• or use pdt (if each provider has one gateway) or mtree+dispatcher

(mtree to translate from a prefix to a dispatcher set id of the provider with many gateways) to send out to providers

If least cost routing is needed towards providers, then you can decide to use one of next modules:

• lcr

• carrierroute

• drouting

Cheers, Daniel

On 12.06.17 21:21, Jean Cérien wrote:

Hello

I am looking at the following configuration:

1/ voip providers, with different external sip servers, will send calls to kamailio, that will dispatch them to asterisk servers, with load balancing & failover. The IP of the asterisk servers are known and fixed.

2/ the asterisk servers will also place calls, going through kamailio, that will dispatch them to the voip provider sip servers, based on R-URI / Prefix.

I've achieved #1 with dispatcher module, I've seen carrierroute module that can achieve #2, but I dont really grasp how to put the two together.

I have the feeling that #1 can also be done with carrierroute, and that I only need to check if the 'from' domain is one my asterisk, and chose a different routing table, but I dont see how to do that

is that a valid approch ? where should I look to do that ?

thanks

J.

---

Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

-- Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

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-- Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

Thanks. I dont have control over the host I need to register to, so I cant unfortunately retrieve the luuid.

Is there a way to lookup in the uacreg table with the IP ? I could only find uac_reg_lookup, but it required the luuid. I've worked around that adding an entry into address, but I am not really satisfied with this method,

Regards, J

On Thu, Jun 15, 2017 at 4:24 AM, Daniel Tryba d.tryba@pocos.nl wrote:

On Wed, Jun 14, 2017 at 05:28:35PM -0400, Jean Cérien wrote:

...

I have the registration working, but when my kamailio receives an

INVITE, I

...

am a bit lost regarding what to do. in my route(AUTH), I am trying to

check

...

uac_reg_status, but no success.

...

and the invite arriving INVITE sip:1100@192.168.2.228 SIP/2.0

Any help will be greatly appreciated

Either accept all INVITEs from the IPs you are registering, use something like db_aliases to map 1100 to your known username or change the setup of the PBX you are registering to to not use the dialed nummer in the R-URI but the username (it is an asterisk so it is the difference between Dial(SIP/${EXTEN}@username) and Dial(SIP/username))

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