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
J'ai aussi ça :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 vérifier si le serveur Astérix arrive bien a communiquer au serveur IPPI :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'.
Code:Host dnsmgr Username Refresh State Reg.Time sip.ippi.com:5060 N Ujonathan 105 Registered Thu, 26 Oct 2017 13:36:09Code: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:
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 responseCode: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
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>
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