You need to engage branch route again in failure route. All those tm
route blocks need to be re-engaged for each t_relay().
Cheers,
Daniel
On 07/01/16 22:09, Daniel W. Graham wrote:
>
> The SDP was updated with RTPProxy IP.
>
>
>
> Yes, config was written around the default config, here are some
> snippets of the config that is related. Do I just need to call branch
> route in the failure route?
>
>
>
> if ($branch(count) > 0) {
>
> t_load_contacts();
>
> t_next_contacts();
>
> t_on_failure("HUNT_FAIL");
>
> }
>
>
>
> route(RELAY);
>
>
>
> ------------------
>
>
>
> route[RELAY] {
>
>
>
> if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
>
> if(!t_is_set("branch_route"))
> t_on_branch("MANAGE_BRANCH");
>
> }
>
> if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
>
> if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY");
>
> }
>
> if (is_method("INVITE")) {
>
> if(!t_is_set("failure_route"))
> t_on_failure("MANAGE_FAILURE");
>
> }
>
>
>
> if (!t_relay()) {
>
> sl_reply_error();
>
> }
>
> exit;
>
> }
>
>
>
> branch_route[MANAGE_BRANCH] {
>
> xlogl("L_INFO", "$ci : New branch [$T_branch_idx] to $ru\n");
>
> route(NATMANAGE);
>
> }
>
>
>
> failure_route["HUNT_FAIL"] {
>
> if (!t_next_contacts()) {
>
> exit;
>
> }
>
>
>
> t_on_failure("HUNT_FAIL");
>
> t_relay();
>
> }
>
> dan-signature
>
>
>
> *From:*Daniel-Constantin Mierla [mailto:mico...@gmail.com]
> *Sent:* Thursday, January 7, 2016 4:24 AM
> *To:* Daniel W. Graham <d...@cmsinter.net>; Kamailio (SER) - Users
> Mailing List <sr-users@lists.sip-router.org>
> *Subject:* Re: [SR-Users] Audio issue when using 2 port ATA
>
>
>
>
>
> On 06/01/16 21:28, Daniel W. Graham wrote:
>
> I did more experimenting and seams the issue only exists in two of
> three configurations. If I can fix the first I think it will fix
> the second as well.
>
>
>
> If both ATA ports share the same username and serial forking is
> used, the issue as described below happens. Looks like the issue
> is that I never called route(NATMANAGE) in the serial forking
> failure route.
>
>
> If you are having your config based on default kamailio.cfg, then you
> should engage the branch route before sending out any invite.
>
> Cheers,
> Daniel
>
>
>
>
> -Dan
>
>
>
> *From:*sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On
> Behalf Of *Daniel W. Graham
> *Sent:* Wednesday, January 6, 2016 3:06 PM
> *To:* mico...@gmail.com <mailto:mico...@gmail.com>; Kamailio (SER)
> - Users Mailing List <sr-users@lists.sip-router.org>
> <mailto:sr-users@lists.sip-router.org>
> *Subject:* Re: [SR-Users] Audio issue when using 2 port ATA
>
>
>
> I do control, this particular setup is in my lab. I just took
> another look at the captures and see both RTP streams (viewing in
> front of firewall). First call rtp is sourced from
> Kamailio(rtpproxy) second call rtp is sourced from one of the
> backend asterisk servers (which is where the issue is, should also
> be from rtpproxy).
>
>
>
> -Dan
>
>
>
> *From:*Daniel-Constantin Mierla [mailto:mico...@gmail.com]
> *Sent:* Wednesday, January 6, 2016 8:09 AM
> *To:* Daniel W. Graham <d...@cmsinter.net
> <mailto:d...@cmsinter.net>>; Kamailio (SER) - Users Mailing List
> <sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org>>
> *Subject:* Re: [SR-Users] Audio issue when using 2 port ATA
>
>
>
> Is the firewall a system that you control and can do traces on it?
> Can you see rtp coming to it? Is it forwarded?
>
> Cheers,
> Daniel
>
> On 06/01/16 13:40, Daniel W. Graham wrote:
>
> Firewall is not doing sip alg, I have compared traces and they
> are the same.
>
> Daniel W. Graham
>
> CMSInter.net <http://cmsinter.net> LLC
>
> 989.400.4230
>
>
> On Jan 6, 2016, at 3:05 AM, Daniel-Constantin Mierla
> <mico...@gmail.com <mailto:mico...@gmail.com>> wrote:
>
> Hello,
>
> is the firewall doing SIP ALG?
>
> Can you get a SIP network trace on UA? If yes, compare it
> with the one captured on server.
>
> Cheers,
> Daniel
>
> On 06/01/16 01:50, Daniel W. Graham wrote:
>
> Setup is -
>
>
>
> 2 port ATA <> FIREWALL <> KAMAILIO / RTPPROXY <> ASTERISK
>
>
>
> If I have a single port in use behind the firewall,
> all NAT functions work properly and media is relayed
> through rtpproxy.
>
>
>
> If I have both ports in use behind the firewall, when
> outbound calls from UA are placed there is two way
> audio on both calls. However if inbound calls are
> placed to UA, the first call works, second call only
> has outbound audio.
>
>
>
> Different SIP URI is used for each port.
>
>
>
> If the firewall is eliminated everything works fine.
>
>
>
> Anyone have an idea how to troubleshoot or what could
> be missing? I have done packet captures on both the UA
> side and Kamailio side, and I see two RTP flows (rtp
> ports match on both sides as well) despite lack of
> inbound audio on the second call.
>
>
>
> If I can post anything config wise that would help let
> me know.
>
>
>
> Thanks!
>
>
>
> -Dan
>
>
>
>
>
>
> _______________________________________________
>
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
> mailing list
>
> sr-users@lists.sip-router.org
> <mailto:sr-users@lists.sip-router.org>
>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
>
> Daniel-Constantin Mierla
>
> http://twitter.com/#!/miconda
> <http://twitter.com/#%21/miconda> -
> http://www.linkedin.com/in/miconda
>
> Book: SIP Routing With Kamailio - http://www.asipto.com
>
> http://miconda.eu
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
> mailing list
> sr-users@lists.sip-router.org
> <mailto:sr-users@lists.sip-router.org>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
>
> Daniel-Constantin Mierla
>
> http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -
> http://www.linkedin.com/in/miconda
>
> Book: SIP Routing With Kamailio - http://www.asipto.com
>
> http://miconda.eu
>
>
>
> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -
> http://www.linkedin.com/in/miconda
> Book: SIP Routing With Kamailio - http://www.asipto.com
> http://miconda.eu
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
http://miconda.eu
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