2 Jan
2018
2 Jan
'18
4:04 p.m.
Hello
Any suggestion as to why the ACK is not forwarded to the Asterisk box and
stays on the kamailio server ?
REgards
J.
On Tue, Dec 26, 2017 at 11:49 AM, Jean Cérien <cerien.jean(a)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(a)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(a)127.0.0.1} 2(20129) ERROR: <script>: =======*
request ACK before relay
{1 2 ACK 1473723014(a)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(a)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/52111974b4571e05
62e8e731df80f48dbc504915
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(a)gmail.com>
wrote:
--
Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
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(a)gmail.com>om>:
>
> 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=z9h
> G4bKjq16fi00d85uee181ic0.1
> To: <sip:kamailioIP:5060>;tag=as7f10ed48
> From: <sip:sourcenumber@orange-multimedia.fr>;tag=SD8u3ob01-7dd8ef
> de-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 Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com