On Tue, Dec 26, 2017 at 11:49 AM, Jean Cérien <cerien.jean@gmail.com> wrote:

Daniel

Many thanks for your help in this holidays season.

The ACK is now going out and executions follows, which is good, however, the ack is sent to the Kamailio box itself and not actually forwarded to the asterisk (I've attached at the end of this message the initial invite, received ACK and "forwarded" ACK).

Both packet go through very classic config, starting with dlg_manage(), then if (has_totag()), if (loose_route()), which leads to route(RELAY), which does the actual call to t_relay.

This sequence works fine with other providers, I cant see what is wrong !

Regards
J

Initial invite received from provider

2017/12/26 15:21:47.414524 VOIP-IP:5060 -> KAMAILIOIP:5060
INVITE sip:CALLED-NUM@KAMAILIOIP SIP/2.0
Via: SIP/2.0/UDP VOIP-IP:5060;branch=z9hG4bKh5r5mp20007mno86oit0.1
P-Asserted-Identity: <sip:CALLING-NUM@orange-multimedia.fr>
To: <sip:CALLED-NUM@KAMAILIOIP:5060>
From: <sip:CALLING-NUM@orange-multimedia.fr>;tag=SDf1f0501-496399c4-0016-096a-0000-0000
Call-ID: SDf1f0501-70980730711ea436a8ed241985dddc7d-v300g00040
CSeq: 1 INVITE
Max-Forwards: 65
Contact: <sip:CALLING-NUM@VOIP-IP:5060;transport=udp>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS
Supported: uui,timer
Accept: application/sdp, application/isup, application/xml
P-Access-Network-Info: GSTN;operator-specific-GI="639720000";network-provided
Content-Type: application/sdp
Content-Length: 221
Session-Expires: 1800; refresher=uac
Min-SE: 90

ACK Received from provider

2017/12/26 15:21:47.473423 VOIP-IP:5060 -> KAMAILIOIP:5060
ACK sip:CALLED-NUM@KAMAILIOIP:5060 SIP/2.0
Via: SIP/2.0/UDP VOIP-IP:5060;branch=z9hG4bKmdfnm110182k1pbvl180.1
To: <sip:KAMAILIOIP:5060>;tag=as3eec8631
From: <sip:CALLING-NUM@orange-multimedia.fr>;tag=SDf1f0501-496399c4-0016-096a-0000-0000
Call-ID: SDf1f0501-70980730711ea436a8ed241985dddc7d-v300g00040
CSeq: 1 ACK
Max-Forwards: 66
Content-Length: 0
Route: <sip:KAMAILIOIP;lr=on;ftag=SDf1f0501-496399c4-0016-096a-0000-0000;did=eea.aa52>

ACK Forwarded

2017/12/26 15:21:47.473898 KAMAILIOIP:5060 -> KAMAILIOIP:5060
ACK sip:KAMAILIOIP;lr=on;ftag=SDf1f0501-496399c4-0016-096a-0000-0000;did=eea.aa52 SIP/2.0
Via: SIP/2.0/UDP KAMAILIOIP;branch=z9hG4bK52bb.2e7f3f0b7c8c60b92bfa2d945cc9829d.0
Via: SIP/2.0/UDP VOIP-IP:5060;branch=z9hG4bKmdfnm110182k1pbvl180.1
To: <sip:KAMAILIOIP:5060>;tag=as3eec8631
From: <sip:CALLING-NUM@orange-multimedia.fr>;tag=SDf1f0501-496399c4-0016-096a-0000-0000
Call-ID: SDf1f0501-70980730711ea436a8ed241985dddc7d-v300g00040
CSeq: 1 ACK
Max-Forwards: 65
Content-Length: 0
On Sat, Dec 23, 2017 at 8:12 AM, Daniel-Constantin Mierla <miconda@gmail.com> wrote:
I just tested and works fine. Try with master branch and default config file.

To test I modified a bit the default kamailio.cfg and I have in route[RELAY]:

    xlog("=======* request $rm before relay\n");
    if (!t_relay()) {
        xlog("=======x request $rm not relayed\n");
        sl_reply_error();
    }
    xlog("=======> request $rm relayed\n");

I get the ACK routed and in logs I get:

{1 2 ACK 1473723014@127.0.0.1} 2(20129) ERROR: <script>: =======* request ACK before relay
{1 2 ACK 1473723014@127.0.0.1} 2(20129) ERROR: <script>: =======> request ACK relayed

Cheers,
Daniel
On 22.12.17 21:00, Jean Cérien wrote:
Many thanks for your help.

I've applied the patch, recompiled and I see no difference unfortunately. The ACK is not forwarded, and execution after t_relay stops.

I've set the debug level to 4 and captured the traces - I believe the section dealing with the ACK is there: https://pastebin.com/TG03YQWC

KR

J.
On Fri, Dec 22, 2017 at 11:59 AM, Daniel-Constantin Mierla <miconda@gmail.com> wrote:
Hello,

can you give a try with latest master branch or by using the patch from next link?

- https://github.com/kamailio/kamailio/commit/52111974b4571e0562e8e731df80f48dbc504915

Apparently the t_realy() stopped script execution (by returning 0) when e2e ACK forwarding was successful. The patch should change that to return true.

If all works fine while testing, then I will backport.

Cheers,
Daniel
On 22.12.17 14:05, Jean Cérien wrote:
Hello
I've tried mhomed = 1 with no luck - not so sure what you mean by default route, I've added listen = udp:myadsress:5060

I also have auto_aliases=no and alias=myaddress

However, what bugs me is that execution seems to stop INSIDE t_relay and the info message after the t_relay is not displayed.

I hate to say this, but could this be a bug ?

Rgds

J.

On Fri, Dec 22, 2017 at 1:52 AM, Sergey Safarov <s.safarov@gmail.com> wrote:

Check that your installation have one NIC with only one default route on host.
If not check that "mhomed=1" is enabled.

Sergey

пт, 22 дек. 2017 г. в 0:02, Jean Cérien <cerien.jean@gmail.com>:

Hello

I am using kamailio 5.0.2, on a debian 9 system.

Everything was running fine, until one of our voip provider changed his switch. Our kamailio is relaying between several voip providers and several asterisk (only the signalisation, no rtp).

When we get an invite from this new switch, we select an asterisk and relay it correctly to this box. However, the OK is relayed back. But when the voip providers sends an ACK for this OK, the t_relay function does not return at all, it just dies with no action and no error.

Here is the snippet from route(relay)

xlog("L_INFO","route(relay) @@ $rm - Source: $si:$sp, fu:$fu, tu:$tu\n" );

$var(restrelay)=t_relay();

xlog("L_INFO","route(relay) @@ $rm - t_relay result: $var(restrelay)" );

if (!$var(restrelay)) {

When processing the initial invite, I do get both INFO messages. When the ACK is processed, I only get one INFO message, and no ACK is relayed - so it seems execution dies in the t_relay

What could be wrong ???

J.

Details of the ACK:

ACK sip:dialednumber@kamailioIP:5060 SIP/2.0

Via: SIP/2.0/UDP VOIPPROVIDERIP:5060;branch=z9hG4bKjq16fi00d85uee181ic0.1

To: <sip:kamailioIP:5060>;tag=as7f10ed48

From: <sip:sourcenumber@orange-multimedia.fr>;tag=SD8u3ob01-7dd8efde-0002-00d5-0000-0000

Call-ID: SD8u3ob01-688d41f5e2164ceeafe76f40b82b3f97-v300g00030

CSeq: 1 ACK

Max-Forwards: 66

Content-Length: 0

Route: <sip:kamailioIP;lr=on;ftag=SD8u3ob01-7dd8efde-0002-00d5-0000-0000;did=a0b.49d>

(should you have enough width; the diagram will look ok)

VOIP PROVIDER KAMAILIO ASTERISK

21:38:54.685149 │ ──────────────────────────> │ │

▒ +0.000451 │ 100 trying -- your call is │ │

▒ 21:38:54.685600 │ <────────────────────────── │ │

▒ +0.000084 │ │ INVITE (SDP) │

▒ 21:38:54.685684 │ │ ──────────────────────────> │

▒ +0.000831 │ │ 100 Trying │

▒ 21:38:54.686515 │ │ <────────────────────────── │

▒ +0.000471 │ │ 200 OK (SDP) │

▒ 21:38:54.686986 │ │ <────────────────────────── │

▒ +0.000394 │ 200 OK (SDP) │ │

▒ 21:38:54.687380 │ <────────────────────────── │ │

▒ +0.038694 │ ACK │ │

▒ 21:38:54.726074 │ ──────────────────────────> │ │

▒ +0.060155 │ │ 200 OK (SDP) │

▒ 21:38:54.786229 │ │ <<<──────────────────────── │

▒ +0.000138 │ 200 OK (SDP) │ │

▒ 21:38:54.786367 │ <<<──────────────────────── │ │

▒ +0.005721 │ ACK │ │

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-- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
-- 
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com