Merry xmas to you all. Best regards atle Den 24. des. 2012 22:04 skrev "Brandon Armstead" brandon@cryy.com følgende:

Merry Christmas!!!

On Mon, Dec 24, 2012 at 1:00 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:

...

Another year getting to its end! Looking back, looks like one with the greatest achievements in the development of the project so far. Year is not done, so I save that summary for one week later, there is still stuff on its way to our source code repository.

Now I want to thank to everyone promoting and contributing to the project, from developers to community members, and wish Merry Christmas and great winter holidays to all supporters of our project!

Cheers, Daniel

-- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Hey Guys...a very Merry Christmas and Happy New Great Year!

Thanks a lot for your work! Il giorno 25/dic/2012 00:12, "Konstantin M." evilzluk@gmail.com ha scritto:

Merry Christmas!!!

2012/12/24 Atle Samuelsen atle.samuelsen@gmail.com

...

Merry xmas to you all. Best regards atle Den 24. des. 2012 22:04 skrev "Brandon Armstead" brandon@cryy.com følgende:

...

Merry Christmas!!!

On Mon, Dec 24, 2012 at 1:00 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:

...

Another year getting to its end! Looking back, looks like one with the greatest achievements in the development of the project so far. Year is not done, so I save that summary for one week later, there is still stuff on its way to our source code repository.

Now I want to thank to everyone promoting and contributing to the project, from developers to community members, and wish Merry Christmas and great winter holidays to all supporters of our project!

Cheers, Daniel

-- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

______________________________**_________________ Kamailio (OpenSER) - Business mailing list Business@lists.kamailio.org http://lists.kamailio.org/cgi-**bin/mailman/listinfo/businesshttp://lists.kamailio.org/cgi-bin/mailman/listinfo/business http://lists.openser-project.**org/cgi-bin/mailman/listinfo/**businesshttp://lists.openser-project.org/cgi-bin/mailman/listinfo/business

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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

17 jan 2013 kl. 18:07 skrev "Scott, Matt" mscott@homeadvisor.com:

Pstn->Kamailio->dispatcher->asterisk

When calls come in, they are sent to asterisk, and the peer is negotiated as Kamailio. How can I have different peer settings, for example dtmf?

I have an Asterisk branch that separates peers beyond the proxy (pinetree-1.4) based on the via received headers. I have it in production for Asterisk 1.4 in multiple places, but don't know the current state in relationship to latest source code.

I’m thinking I want the call to come in, then when the invite is sent to asterisk, the invite should be from the pstn-peer and not Kamailio, make sense? Then I can have separate peers, with their own dtmf/audio settings, but I see the invite as:

Remember that Asterisk first match incoming calls on users with the FROM header. You can separate based on the From header. But for PSTN, that doesn't really help...

If you have a limited set of profiles you can add port numbers to Asterisk peers, like have a peer on port 5060 and another on the kamailio IP on 5065. Have kamailio open both sockets and use force_socket to send from the proper one.

/O

INVITE sip:+1866NXXNXXX@<kamailio ip>:5060 SIP/2.0 Via: SIP/2.0/UDP <Kamailio ip>;branch=z9hG4bKc087.12db3c91.0 Via: SIP/2.0/UDP <pstn-peer ip>:5060;branch=z9hG4bK0cB824c55100a4f2b8b From: <sip:+1NXXNXXXXXXXX@<pstn-peer ip>:5060;isup-oli=0>;tag=gK0c731df8 To: <sip:+1NXXNXXNXXX@<kamailio ip>:5060> Call-ID: 1024240378_131584110@<pstn-peer ip>

Any help or suggestions are very welcome..

Matt Scott

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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

Let's see if I can do this, so something like this? Using the PSTN Source IP to decide on the asterisk peer? Not really sure on what variable dispatch will return as it's chosen ip?

IP_auth ->Dispatch->Force Socket->Relay

route{

if (!allow_source_address("1")) { sl_send_reply("403", "Forbidden"); exit; }; # load-balance dispatching on gateways group '1' if(!ds_select_dst("1", "10")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n");

t_on_failure("RTF_DISPATCH");

if ($si == "1.2.3.1") { force_send_socket($sndto:5065); } else if ($si == "1.2.3.2") { force_send_socket($sndto:5070); } t_relay();

return;

Most definitely, thank you for your time.

Matt Scott

From: sr-users-bounces@lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Olle E. Johansson Sent: Friday, January 18, 2013 1:09 AM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] Invite - negotiate to asterisk as peer not kamailio

17 jan 2013 kl. 18:07 skrev "Scott, Matt" <mscott@homeadvisor.commailto:mscott@homeadvisor.com>:

Pstn->Kamailio->dispatcher->asterisk

When calls come in, they are sent to asterisk, and the peer is negotiated as Kamailio. How can I have different peer settings, for example dtmf? I have an Asterisk branch that separates peers beyond the proxy (pinetree-1.4) based on the via received headers. I have it in production for Asterisk 1.4 in multiple places, but don't know the current state in relationship to latest source code.

I'm thinking I want the call to come in, then when the invite is sent to asterisk, the invite should be from the pstn-peer and not Kamailio, make sense? Then I can have separate peers, with their own dtmf/audio settings, but I see the invite as:

Remember that Asterisk first match incoming calls on users with the FROM header. You can separate based on the From header. But for PSTN, that doesn't really help...

If you have a limited set of profiles you can add port numbers to Asterisk peers, like have a peer on port 5060 and another on the kamailio IP on 5065. Have kamailio open both sockets and use force_socket to send from the proper one.

/O

INVITE sip:+1866NXXNXXX@<kamailiosip:+1866NXXNXXX@%3ckamailio ip>:5060 SIP/2.0 Via: SIP/2.0/UDP <Kamailio ip>;branch=z9hG4bKc087.12db3c91.0 Via: SIP/2.0/UDP <pstn-peer ip>:5060;branch=z9hG4bK0cB824c55100a4f2b8b From: <sip:+1NXXNXXXXXXXX@<pstn-peer ip>:5060;isup-oli=0>;tag=gK0c731df8 To: <sip:+1NXXNXXNXXX@<kamailio ip>:5060> Call-ID: 1024240378_131584110@<pstn-peer ip>

Any help or suggestions are very welcome..

Matt Scott

_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.orgmailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users