Alors j’ai effectué les modifications que tu m’as indiquées, quand j’appelle ça raccroche directement j’ai aussi un nombre considérable de lignes qui apparaisse plus rapidement, quand temps normal
Code:
[Oct 26 13:27:37] NOTICE[3764]: chan_sip.c:28627 handle_request_register: Registration from '"6003" <sip:6003@90.32.145.143>' failed for '153.120.82.250:5310' - Wrong password
J'ai aussi ça :
Code:
[Oct 26 13:27:38] NOTICE[3764][C-0000000a]: chan_sip.c:26401 handle_request_invite: Call from '' (158.69.250.70:5074) to extension '44080048833701703' rejected because extension not found in context 'default'.
J'ai vérifier si le serveur Astérix arrive bien a communiquer au serveur IPPI :
Code:
Host dnsmgr Username Refresh State Reg.Time
sip.ippi.com:5060 N Ujonathan 105 Registered Thu, 26 Oct 2017 13:36:09
Code:
SRV-VOIP*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
6001/jonathan (Unspecified) D Auto (No) No 0 Unmonitored
6002/kevin (Unspecified) D Auto (No) No 0 Unmonitored
ippi_entrant_sortant/Ujon 194.169.214.30 Yes Yes 5060 Unmonitored
my_phone/my_phone 192.168.1.113 D No No 54983 Unmonitored
Du cotée X-lite je suis connecter avec le compte my_phone sur mon serveur qui a une adresse ip 192.168.1.58
Du cotée Modem voici les ouvertures de port vers mon serveur Astérix:
![](https://img4.hostingpics.net/pics/535470asterixnat.png)
J'ai été vérifier la fameuse erreur contexte peer .
Code:
SRV-VOIP*CLI> dialplan show default
[ Context 'default' created by 'pbx_config' ]
'6001' => hint: SIP/6001&IAX2/6001 [pbx_config]
1. Dial(${HINT}) [pbx_config]
'6002' => hint: SIP/6002&IAX2/6002 [pbx_config]
1. Dial(${HINT}) [pbx_config]
'6003' => hint: SIP/6003&IAX2/6003 [pbx_config]
1. Dial(${HINT}) [pbx_config]
Code:
SRV-VOIP*CLI> dialplan reload
Dialplan reloaded.
WARNING[3764]: chan_sip.c:4072 retrans_pkt: Retransmission timeout reached on transmission 72cdf7be123719955e49226e5090be0e for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
Code:
SRV-VOIP*CLI> reload
[Oct 26 13:54:28] NOTICE[3764]: chan_sip.c:28627 handle_request_register: Registration from '"6003" <sip:6003@90.32.145.143>' failed for '153.120.82.250:5310' - Wrong password
[Oct 26 13:54:28] WARNING[3810]: res_phoneprov.c:1231 get_defaults: Unable to find a valid server address or name.
[Oct 26 13:54:28] NOTICE[3810]: chan_skinny.c:8441 config_load: Configuring skinny from skinny.conf
[Oct 26 13:54:28] NOTICE[3764]: chan_sip.c:28627 handle_request_register: Registration from '"60" <sip:60@90.32.145.143>' failed for '153.120.82.250:5510' - Wrong password
[Oct 26 13:54:28] NOTICE[3810]: cel_custom.c:97 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
J’essaye de vous mettre un maximum de détails ![Redface](images/smilies/redface.png)
J'ai redémarré mon serveur et est désactiver tous les compte local SIP je n'ai plus de message d'erreur "password Wrong" par contre j'ai ça :
Code:
root@SRV-VOIP:~# asterisk -r
Asterisk 13.17.2, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 13.17.2 currently running on SRV-VOIP (pid = 1113)
SRV-VOIP*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
sip.ippi.com:5060 N Ujonathan 105 Registered Thu, 26 Oct 2017 14:13:21
1 SIP registrations.
SRV-VOIP*CLI>
SRV-VOIP*CLI>
SRV-VOIP*CLI>
SRV-VOIP*CLI>
SRV-VOIP*CLI>
SRV-VOIP*CLI>
SRV-VOIP*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
ippi_entrant_sortant/Ujon 194.169.214.30 Yes Yes 5060 Unmonitored
my_phone/my_phone 192.168.1.113 D No No 54983 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
[Oct 26 14:14:42] NOTICE[1225][C-00000000]: chan_sip.c:26401 handle_request_invite: Call from '' (62.210.203.106:49280) to extension '0046812400432' rejected because extension not found in context 'default'.
[Oct 26 14:14:50] NOTICE[1225]: chan_sip.c:28404 handle_request_subscribe: Received SIP subscribe for peer without mailbox: my_phone
[Oct 26 14:15:02] NOTICE[1225][C-00000001]: chan_sip.c:26401 handle_request_invite: Call from 'my_phone' (192.168.1.113:55120) to extension '06736684' rejected because extension not found in context 'peer'.
[Oct 26 14:15:09] NOTICE[1225][C-00000002]: chan_sip.c:26401 handle_request_invite: Call from '' (158.69.250.70:5070) to extension '99900110048833701703' rejected because extension not found in context 'default'.
[Oct 26 14:15:14] WARNING[1225]: chan_sip.c:4072 retrans_pkt: Retransmission timeout reached on transmission 112663629-73517601-606133390 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Oct 26 14:15:22] NOTICE[1225][C-00000003]: chan_sip.c:26401 handle_request_invite: Call from '' (188.138.9.254:5071) to extension '000972599604681' rejected because extension not found in context 'default'.
SRV-VOIP*CLI>
Vérification dialplan show
Code:
SRV-VOIP*CLI> dialplan show default
There is no existence of 'default' context
Command 'dialplan show default' failed.
[Oct 26 14:32:04] NOTICE[1225][C-0000000c]: chan_sip.c:26401 handle_request_invite: Call from '' (62.210.203.106:64653) to extension '0046812400432' rejected because extension not found in context 'default'.
SRV-VOIP*CLI>
Et enfin voilà ce qui se passe du coter Astérix quand j’appelle avec le compte my_phone sur un téléphone portable:
Code:
Connected to Asterisk 13.17.2 currently running on SRV-VOIP (pid = 1113)
[Oct 26 14:41:38] WARNING[1225]: chan_sip.c:4072 retrans_pkt: Retransmission timeout reached on transmission 656643972-1045403765-482509620 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Oct 26 14:41:44] NOTICE[1225][C-00000006]: chan_sip.c:26401 handle_request_invite: Call from 'my_phone' (192.168.1.113:59180) to extension '06736684' rejected because extension not found in context 'peer'.
Code:
Connected to Asterisk 13.17.2 currently running on SRV-VOIP (pid = 1120)
[Oct 26 15:19:19] WARNING[1227][C-00000007]: chan_sip.c:24003 handle_response_invite: Received response: "Forbidden" from '<sip:my_phone@sip.ippi.com>;tag=as02b4d7cc'
SRV-VOIP*CLI>