[SR-Users] t_relay dying ?
Wilkins, Steve
swwilkins at mitre.org
Tue Jan 2 14:19:00 CET 2018
Hello,
Have you done a trace tcpdump/Wireshark? What does the traffic look like? Are Asterisk and Kamailio on different Servers?
-Steve
From: sr-users [mailto:sr-users-bounces at lists.kamailio.org] On Behalf Of Jean Cérien
Sent: Tuesday, January 2, 2018 8:04 AM
To: Daniel-Constantin Mierla <miconda at gmail.com>
Cc: Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
Subject: Re: [SR-Users] t_relay dying ?
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 at gmail.com<mailto:cerien.jean at 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 at KAMAILIOIP SIP/2.0
Via: SIP/2.0/UDP VOIP-IP:5060;branch=z9hG4bKh5r5mp20007mno86oit0.1
P-Asserted-Identity: <sip:CALLING-NUM at orange-multimedia.fr<mailto:sip%3ACALLING-NUM at orange-multimedia.fr>>
To: <sip:CALLED-NUM at KAMAILIOIP:5060>
From: <sip:CALLING-NUM at orange-multimedia.fr<mailto:sip%3ACALLING-NUM at 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 at 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 at 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 at orange-multimedia.fr<mailto:sip%3ACALLING-NUM at 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 at orange-multimedia.fr<mailto:sip%3ACALLING-NUM at 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 at gmail.com<mailto:miconda at 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 at 127.0.0.1<mailto:1473723014 at 127.0.0.1>} 2(20129) ERROR: <script>: =======* request ACK before relay
{1 2 ACK 1473723014 at 127.0.0.1<mailto:1473723014 at 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 at gmail.com<mailto:miconda at 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 at gmail.com<mailto:s.safarov at 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 at gmail.com<mailto:cerien.jean at 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 at kamailioIP:5060 SIP/2.0
Via: SIP/2.0/UDP VOIPPROVIDERIP:5060;branch=z9hG4bKjq16fi00d85uee181ic0.1
To: <sip:kamailioIP:5060>;tag=as7f10ed48
From: <sip:sourcenumber at orange-multimedia.fr<mailto:sip%3Asourcenumber at 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<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda>
Kamailio Advanced Training - www.asipto.com<http://www.asipto.com>
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com<http://www.kamailioworld.com>
--
Daniel-Constantin Mierla
www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda>
Kamailio Advanced Training - www.asipto.com<http://www.asipto.com>
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com<http://www.kamailioworld.com>
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