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Discussion: [RÉSOLU] Problème d'appel sortant mais pas entrant.

  1. #1
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    Exclamation [RÉSOLU] Problème d'appel sortant mais pas entrant.

    Salutations !

    Suite à une mise à jour hasardeuse mon Asterisk est passé de la 1.8 à la version 11 ! Grâce aux personnes de ce forum j'ai pu rétablir en grande partie la situation mais il reste un point noir.

    J'ai un soucis d'appel sortant ! Et uniquement sortant car les appels entrant eux fonctionnent. J'avais eu le soucis une fois et dans la configuration sip j'avais du rajouter des infos sur mon réseau interne et l'IP externe de ma box.

    Sauf que là ça n'aide plus donc je pense qu'il y a eu un changement ou que la version 11 a besoin de plus de précisoin. J'avoue ma faiblesse sur la partie réseau. Si je comprend bien un paquet de retour n'arrive pas à bon port et donc la communication ne peut se faire. Comment préciser qu'elle est la bonne passerelle/chemin réseau ? Ça doit être un truc tout bête comme mon ancien problème mais la je cale ... Je précise que l'on a plusieurs Box redondé derrière une machine faisant office de routeur. Lorsque j'avais précisé la partie réseau interne et externe cela fonctionnait bie navec la 1.8 mais plus la 11.

    Merci pour vos lumières.

    Les erreurs:
    Code:
    [Mar 13 11:49:03] WARNING[18377]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission 7e5d60197466d01017c61953018565d1@80.12.90.15:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6400ms with no response
    [Mar 13 11:49:03] WARNING[18377]: chan_sip.c:4204 retrans_pkt: Hanging up call 7e5d60197466d01017c61953018565d1@80.12.90.15:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
      == Everyone is busy/congested at this time (1:0/0/1)
        -- Auto fallthrough, channel 'SIP/34-000001e8' status is 'CHANUNAVAIL'
    SIP.CONF, la partie concernant la configuration général et non les comptes:
    Code:
    ; General Configuration:
    [general]
    defaultexpirey = 1800
    dtmfmode = rfc2833
    ;canreinvite=no
    qualify = yes
    context = others
    port = 5060
    bindaddr = 0.0.0.0
    ;bindaddr = 172.16.1.92
    srvlookup=yes
    limitonpeer = yes
    disallow=all
    allow=ulaw
    allow=alaw
    ;allow=g729
    nat=yes
    ;nat=force_rport,comedia
    ;externip=80.10.115.228
    externip=80.12.90.15
    ;externip=109.26.2.198
    localnet=172.16.1.0/255.255.255.0
    Dernière modification par motarion ; 14/03/2014 à 19h01.

  2. #2
    Membre Association Avatar de quintana
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    Tu pourrais mettre plus d'infos de la CLI stp. Je ne vois pas trop ce qui pourrait empêcher le sortant entre 1.8 et 11. D'ailleurs c'est mieux que tu sois en 11
    Découvrez Wazo sous licence GPLv3 et accessible pour tous : http://www.wazo.community
    Blog Wazo : http://blog.wazo.community
    Wazo est un fork de XiVO.
    Suivez moi sur Twitter !

  3. #3
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    Hm ok pour mettre plus d'info du CLI. En lançant des commandes spécifiques? Ou les logs qui s'affichent naturellement sur le CLI ?

    Je vais laisser le CLI dérouler et checker les logs pour avoir plus de trucs à copier coller si ça peut aider.

    Pour ce qui est de la version 11 je pense qu'ils améliorent les choses donc ce sera pas un mal d'y passer, de toute façon à bosser en testing (pas le choix) sur debian fallait s'attendre à changer de version!

    CLI plus causante:
    Code:
    Privilege escalation protection disabled!
    See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
    Asterisk 11.7.0~dfsg-1+b1, Copyright (C) 1999 - 2013 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 11.7.0~dfsg-1+b1 currently running on ipbx-v2 (pid = 18344)
    [Mar 13 14:15:54] NOTICE[18377]: chan_sip.c:27783 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 50
    [Mar 13 14:18:55] NOTICE[18377]: chan_sip.c:27783 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 50
      == Spawn extension (standard, 2, 1) exited non-zero on 'SIP/ovh-in-000001fb'
    [Mar 13 14:21:56] NOTICE[18377]: chan_sip.c:27783 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 50
      == Using SIP RTP CoS mark 5
        -- Executing [s@standard:1] Answer("SIP/ovh-in-00000204", "") in new stack
        -- Executing [s@standard:2] Set("SIP/ovh-in-00000204", "CALLERID(name)=") in new stack
        -- Executing [s@standard:3] BackGround("SIP/ovh-in-00000204", "01_Eulerian_Tech_Bonjour") in new stack
        -- <SIP/ovh-in-00000204> Playing '01_Eulerian_Tech_Bonjour.ulaw' (language 'fr')
           > 0x245a0e0 -- Probation passed - setting RTP source address to 91.121.129.143:33114
        -- Executing [s@standard:4] BackGround("SIP/ovh-in-00000204", "02_Eulerian_Service_Commercial") in new stack
        -- <SIP/ovh-in-00000204> Playing '02_Eulerian_Service_Commercial.ulaw' (language 'fr')
        -- Executing [s@standard:5] BackGround("SIP/ovh-in-00000204", "03_Eulerian_Menu_1") in new stack
        -- <SIP/ovh-in-00000204> Playing '03_Eulerian_Menu_1.ulaw' (language 'fr')
        -- Executing [s@standard:6] Wait("SIP/ovh-in-00000204", "0.5") in new stack
        -- Executing [s@standard:7] BackGround("SIP/ovh-in-00000204", "04_Eulerian_Service_Account_Management") in new stack
        -- <SIP/ovh-in-00000204> Playing '04_Eulerian_Service_Account_Management.ulaw' (language 'fr')
        -- Executing [s@standard:8] BackGround("SIP/ovh-in-00000204", "05_Eulerian_Menu_2") in new stack
        -- <SIP/ovh-in-00000204> Playing '05_Eulerian_Menu_2.ulaw' (language 'fr')
        -- Executing [s@standard:9] Wait("SIP/ovh-in-00000204", "0.5") in new stack
        -- Executing [s@standard:10] BackGround("SIP/ovh-in-00000204", "06_Eulerian_Service_Marketing") in new stack
        -- <SIP/ovh-in-00000204> Playing '06_Eulerian_Service_Marketing.ulaw' (language 'fr')
        -- Executing [s@standard:11] BackGround("SIP/ovh-in-00000204", "07_Eulerian_Menu_3") in new stack
        -- <SIP/ovh-in-00000204> Playing '07_Eulerian_Menu_3.ulaw' (language 'fr')
        -- Executing [2@standard:1] Queue("SIP/ovh-in-00000204", "support,nTtr,,,") in new stack
      == Using SIP RTP CoS mark 5
        -- Called SIP/23
    --- SNIP ---
        -- SIP/23-00000205 connected line has changed. Saving it until answer for SIP/ovh-in-00000204
        -- SIP/26-0000020c is ringing
        -- SIP/22-0000020b is ringing
        -- SIP/24-00000207 is ringing
        -- SIP/25-0000020a is ringing
        -- SIP/21-00000208 is ringing
        -- SIP/25-0000020a connected line has changed. Saving it until answer for SIP/ovh-in-00000204
        -- SIP/25-0000020a answered SIP/ovh-in-00000204
           > 0x7f7bd00feb00 -- Probation passed - setting RTP source address to 172.16.1.113:5012
    [Mar 13 14:24:56] NOTICE[18377]: chan_sip.c:27783 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 50
      == Using SIP RTP CoS mark 5
        -- Executing [s@standard:1] Answer("SIP/ovh-in-0000020d", "") in new stack
        -- Executing [s@standard:2] Set("SIP/ovh-in-0000020d", "CALLERID(name)=") in new stack
        -- Executing [s@standard:3] BackGround("SIP/ovh-in-0000020d", "01_Eulerian_Tech_Bonjour") in new stack
        -- <SIP/ovh-in-0000020d> Playing '01_Eulerian_Tech_Bonjour.ulaw' (language 'fr')
           > 0x247ba10 -- Probation passed - setting RTP source address to 91.121.129.143:34100
        -- Executing [2@standard:1] Queue("SIP/ovh-in-0000020d", "support,nTtr,,,") in new stack
      == Using SIP RTP CoS mark 5
    --- snip ---
      == Using SIP RTP CoS mark 5
        -- Called SIP/26
        -- SIP/26-00000215 connected line has changed. Saving it until answer for SIP/ovh-in-0000020d
        -- SIP/22-00000214 connected line has changed. Saving it until answer for SIP/ovh-in-0000020d
        -- SIP/28-00000212 connected line has changed. Saving it until answer for SIP/ovh-in-0000020d
        -- SIP/21-00000211 connected line has changed. Saving it until answer for SIP/ovh-in-0000020d
        -- SIP/24-00000210 connected line has changed. Saving it until answer for SIP/ovh-in-0000020d
        -- SIP/27-0000020f connected line has changed. Saving it until answer for SIP/ovh-in-0000020d
        -- SIP/23-0000020e connected line has changed. Saving it until answer for SIP/ovh-in-0000020d
        -- SIP/24-00000210 is ringing
        -- SIP/26-00000215 is ringing
        -- SIP/21-00000211 is ringing
        -- SIP/22-00000214 is ringing
        -- SIP/26-00000215 connected line has changed. Saving it until answer for SIP/ovh-in-0000020d
        -- SIP/26-00000215 answered SIP/ovh-in-0000020d
           > 0x7f7bd4034f30 -- Probation passed - setting RTP source address to 172.16.1.90:5020
        -- Got SIP response 480 "Temporarily not available" back from 172.16.1.64:5060
        -- Got SIP response 480 "Temporarily not available" back from 172.16.1.106:5060
    [Mar 13 14:27:56] NOTICE[18377]: chan_sip.c:27783 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 50
      == Spawn extension (standard, 2, 1) exited non-zero on 'SIP/ovh-in-0000020d'
      == Using SIP RTP CoS mark 5
        -- Executing [s@standard:1] Answer("SIP/ovh-in-00000216", "") in new stack
        -- Executing [s@standard:2] Set("SIP/ovh-in-00000216", "CALLERID(name)=") in new stack
        -- Executing [s@standard:3] BackGround("SIP/ovh-in-00000216", "01_Eulerian_Tech_Bonjour") in new stack
        -- <SIP/ovh-in-00000216> Playing '01_Eulerian_Tech_Bonjour.ulaw' (language 'fr')
           > 0x2495fd0 -- Probation passed - setting RTP source address to 91.121.129.142:30748
        -- Executing [s@standard:4] BackGround("SIP/ovh-in-00000216", "02_Eulerian_Service_Commercial") in new stack
        -- <SIP/ovh-in-00000216> Playing '02_Eulerian_Service_Commercial.ulaw' (language 'fr')
        -- Executing [s@standard:5] BackGround("SIP/ovh-in-00000216", "03_Eulerian_Menu_1") in new stack
        -- <SIP/ovh-in-00000216> Playing '03_Eulerian_Menu_1.ulaw' (language 'fr')
        -- Executing [s@standard:6] Wait("SIP/ovh-in-00000216", "0.5") in new stack
        -- Executing [s@standard:7] BackGround("SIP/ovh-in-00000216", "04_Eulerian_Service_Account_Management") in new stack
        -- <SIP/ovh-in-00000216> Playing '04_Eulerian_Service_Account_Management.ulaw' (language 'fr')
        -- Executing [s@standard:8] BackGround("SIP/ovh-in-00000216", "05_Eulerian_Menu_2") in new stack
        -- <SIP/ovh-in-00000216> Playing '05_Eulerian_Menu_2.ulaw' (language 'fr')
        -- Executing [1@standard:1] Queue("SIP/ovh-in-00000216", "sales,nTtr,,,") in new stack
      == Using SIP RTP CoS mark 5
        -- Called SIP/17
    --- SNIP ---
        -- SIP/17-00000217 connected line has changed. Saving it until answer for SIP/ovh-in-00000216
        -- SIP/11-00000219 is ringing
        -- SIP/16-0000021c is ringing
        -- SIP/17-00000217 is ringing
        -- SIP/12-0000021b is ringing
        -- SIP/14-00000218 is ringing
        -- SIP/15-0000021a is ringing
        -- SIP/17-00000217 connected line has changed. Saving it until answer for SIP/ovh-in-00000216
        -- SIP/17-00000217 answered SIP/ovh-in-00000216
           > 0x7f7bd40008f0 -- Probation passed - setting RTP source address to 172.16.1.105:5018
    [Mar 13 14:30:56] NOTICE[18377]: chan_sip.c:27783 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 50
      == Spawn extension (standard, 2, 1) exited non-zero on 'SIP/ovh-in-00000204'
      == Spawn extension (standard, 1, 1) exited non-zero on 'SIP/ovh-in-00000216'
      == Using SIP RTP CoS mark 5
        -- Executing [00298337600@users:1] Dial("SIP/13-0000021d", "SIP/0298337600@ovh-out,,r") in new stack
      == Using SIP RTP CoS mark 5
        -- Called SIP/0298337600@ovh-out
    [Mar 13 14:33:53] WARNING[18377]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission 47eec1c34226fe533585c1e423e02ebf@80.12.90.15:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6400ms with no response
    [Mar 13 14:33:53] WARNING[18377]: chan_sip.c:4204 retrans_pkt: Hanging up call 47eec1c34226fe533585c1e423e02ebf@80.12.90.15:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
      == Everyone is busy/congested at this time (1:0/0/1)
        -- Auto fallthrough, channel 'SIP/13-0000021d' status is 'CHANUNAVAIL'
    [Mar 13 14:33:57] NOTICE[18377]: chan_sip.c:27783 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 50

  4. #4
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    Addendum avec cette fois des tentatives d'appels sortant avec toujours le même soucis.

    Code:
        -- Executing [00298337600@users:1] Dial("SIP/13-0000021d", "SIP/0298337600@ovh-out,,r") in new stack
      == Using SIP RTP CoS mark 5
        -- Called SIP/0298337600@ovh-out
    [Mar 13 14:33:53] WARNING[18377]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission 47eec1c34226fe533585c1e423e02ebf@80.12.90.15:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6400ms with no response
    [Mar 13 14:33:53] WARNING[18377]: chan_sip.c:4204 retrans_pkt: Hanging up call 47eec1c34226fe533585c1e423e02ebf@80.12.90.15:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
      == Everyone is busy/congested at this time (1:0/0/1)
        -- Auto fallthrough, channel 'SIP/13-0000021d' status is 'CHANUNAVAIL'
    [Mar 13 14:33:57] NOTICE[18377]: chan_sip.c:27783 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 50
    [Mar 13 14:36:57] NOTICE[18377]: chan_sip.c:27783 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 50
      == Using SIP RTP CoS mark 5
        -- Executing [s@standard:1] Answer("SIP/ovh-in-0000021f", "") in new stack
        -- Executing [s@standard:2] Set("SIP/ovh-in-0000021f", "CALLERID(name)=") in new stack
        -- Executing [s@standard:3] BackGround("SIP/ovh-in-0000021f", "01_Eulerian_Tech_Bonjour") in new stack
        -- <SIP/ovh-in-0000021f> Playing '01_Eulerian_Tech_Bonjour.ulaw' (language 'fr')
           > 0x2452b10 -- Probation passed - setting RTP source address to 91.121.129.142:34890
        -- Executing [s@standard:4] BackGround("SIP/ovh-in-0000021f", "02_Eulerian_Service_Commercial") in new stack
        -- <SIP/ovh-in-0000021f> Playing '02_Eulerian_Service_Commercial.ulaw' (language 'fr')
        -- Executing [s@standard:5] BackGround("SIP/ovh-in-0000021f", "03_Eulerian_Menu_1") in new stack
        -- <SIP/ovh-in-0000021f> Playing '03_Eulerian_Menu_1.ulaw' (language 'fr')
        -- Executing [s@standard:6] Wait("SIP/ovh-in-0000021f", "0.5") in new stack
        -- Executing [s@standard:7] BackGround("SIP/ovh-in-0000021f", "04_Eulerian_Service_Account_Management") in new stack
        -- <SIP/ovh-in-0000021f> Playing '04_Eulerian_Service_Account_Management.ulaw' (language 'fr')
        -- Executing [1@standard:1] Queue("SIP/ovh-in-0000021f", "sales,nTtr,,,") in new stack
      == Using SIP RTP CoS mark 5
        -- Called SIP/17
      == Using SIP RTP CoS mark 5
        -- Called SIP/14
      == Using SIP RTP CoS mark 5
        -- Called SIP/11
      == Using SIP RTP CoS mark 5
        -- Called SIP/15
      == Using SIP RTP CoS mark 5
        -- Called SIP/12
      == Using SIP RTP CoS mark 5
        -- Called SIP/16
        -- SIP/16-00000225 connected line has changed. Saving it until answer for SIP/ovh-in-0000021f
        -- SIP/12-00000224 connected line has changed. Saving it until answer for SIP/ovh-in-0000021f
        -- SIP/15-00000223 connected line has changed. Saving it until answer for SIP/ovh-in-0000021f
        -- SIP/11-00000222 connected line has changed. Saving it until answer for SIP/ovh-in-0000021f
        -- SIP/14-00000221 connected line has changed. Saving it until answer for SIP/ovh-in-0000021f
        -- SIP/17-00000220 connected line has changed. Saving it until answer for SIP/ovh-in-0000021f
        -- SIP/11-00000222 is ringing
        -- SIP/16-00000225 is ringing
        -- SIP/14-00000221 is ringing
        -- SIP/12-00000224 is ringing
        -- SIP/15-00000223 is ringing
        -- SIP/17-00000220 is ringing
        -- SIP/17-00000220 connected line has changed. Saving it until answer for SIP/ovh-in-0000021f
        -- SIP/17-00000220 answered SIP/ovh-in-0000021f
           > 0x7f7bd40a7b00 -- Probation passed - setting RTP source address to 172.16.1.105:5020
    [Mar 13 14:39:57] NOTICE[18377]: chan_sip.c:27783 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 50
    [Mar 13 14:42:58] NOTICE[18377]: chan_sip.c:27783 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 50
      == Using SIP RTP CoS mark 5
        -- Executing [00603357850@users:1] Dial("SIP/34-00000226", "SIP/0603357850@ovh-out,,r") in new stack
      == Using SIP RTP CoS mark 5
        -- Called SIP/0603357850@ovh-out
    [Mar 13 14:43:19] WARNING[18377]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission 447cc01e731ee18a07abcd212a5612a4@80.12.90.15:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6401ms with no response
    [Mar 13 14:43:19] WARNING[18377]: chan_sip.c:4204 retrans_pkt: Hanging up call 447cc01e731ee18a07abcd212a5612a4@80.12.90.15:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
      == Everyone is busy/congested at this time (1:0/0/1)
        -- Auto fallthrough, channel 'SIP/34-00000226' status is 'CHANUNAVAIL'

  5. #5
    Membre Association Avatar de quintana
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    Es-tu enregistré sur OVH, genre un sip show peers et un sip show registry serait pas mal. As tu activé le qualify aussi vu que tu es natté ? En tout cas je dirai en regardant comme cela que ça ne vient pas d'Asterisk directement. Tu dois avoir un autre soucis plus orienté réseau ou juste OVH dans le choux en même temps Là y a pas d'erreur juste une tentative d'aller sur le serveur OVH mais ça bloque. Tu n'as pas un firewall qui te bloquerai ?
    Découvrez Wazo sous licence GPLv3 et accessible pour tous : http://www.wazo.community
    Blog Wazo : http://blog.wazo.community
    Wazo est un fork de XiVO.
    Suivez moi sur Twitter !

  6. #6
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    Oui je suis chez OVH et le qualify est à YES dans la conf SIP General donc tous devraient l'avoir.

    J'ai 2 comptes SIP OVH, dont un qui est totalement commenté et ne sert QUE pour les tests. Par contre avec les commande SIP, je ne comprend pas pourquoi j'ai deux username différent ... Je viens juste de remarquer en voulant les masquer. Doit etre ce que renvoi OVH pour le register.

    Quant au Firewall normalement on en a pas. Pas moi qui m'occupe de cette partie, mais on avait le routeur qui forcait le passage SIP sur une Box spécifique mais cela a été retiré.

    Je viens de voir aussi que pour le nat dans la configuration OVH j'avais force_rport,comedia donc je l'ai repassé à yes comme c'était avant.

    Maintenant je sais d'où viennent mes cheveux blanc !

    Commandes SIP:
    Code:
    ipbx-v2*CLI> sip show registry
    Host                                    dnsmgr Username       Refresh State                Reg.Time                 
    sip.ovh.fr:5060                         N      003397229902      1785 Registered           Thu, 13 Mar 2014 15:17:02
    1 SIP registrations.
    ipbx-v2*CLI> sip show peers
    Name/username             Host                                    Dyn Forcerport ACL Port     Status      Description                      
    11/11                     172.16.1.245                             D   N             5060     OK (18 ms)                                   
    12/12                     172.16.1.97                              D   N             5060     OK (74 ms)                                   
    13/13                     172.16.1.84                              D   N             5060     OK (77 ms)                                   
    14/14                     172.16.1.83                              D   N             5060     OK (103 ms)                                  
    15/15                     172.16.1.102                             D   N             5060     OK (104 ms)                                  
    16/16                     172.16.1.104                             D   N             5060     OK (30 ms)                                   
    17/17                     172.16.1.105                             D   N             5060     OK (31 ms)                                   
    21/21                     172.16.1.87                              D   N             5060     OK (32 ms)                                   
    22/22                     172.16.1.88                              D   N             5060     OK (32 ms)                                   
    23/23                     172.16.1.138                             D   N             5060     OK (105 ms)                                  
    24/24                     172.16.1.82                              D   N             5060     OK (31 ms)                                   
    25/25                     172.16.1.113                             D   N             5060     OK (19 ms)                                   
    26/26                     172.16.1.90                              D   N             5060     OK (75 ms)                                   
    27/27                     172.16.1.64                              D   N             5060     OK (78 ms)                                   
    28/28                     172.16.1.106                             D   N             5060     OK (102 ms)                                  
    31/31                     172.16.1.108                             D   N             5060     OK (31 ms)                                   
    32                        (Unspecified)                            D   N             0        UNKNOWN                                      
    33/33                     172.16.1.107                             D   N             5060     OK (19 ms)                                   
    34/34                     172.16.1.85                              D   N             5060     OK (104 ms)                                  
    35/35                     172.16.1.103                             D   N             5060     OK (104 ms)                                  
    36                        (Unspecified)                            D   N             0        UNKNOWN                                      
    41/41                     172.16.1.140                             D   N             5060     OK (22 ms)                                   
    42/42                     172.16.1.141                             D   N             5060     OK (20 ms)                                   
    43/43                     172.16.1.142                             D   N             5060     OK (21 ms)                                   
    50/50                     172.16.1.56                              D   N             52062    OK (105 ms)                                  
    ovh-in                    91.121.129.20                                N             5060     OK (3 ms)                                    
    ovh-out/0033972299021     91.121.129.20                                N             5060     OK (10 ms)                                   
    27 sip peers [Monitored: 25 online, 2 offline Unmonitored: 0 online, 0 offline]

  7. #7
    Membre Association Avatar de quintana
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    En tout cas je ne vois pas de souci côté Asterisk avec ce que tu donnes, à mon avis il faut que tu fasses du tcpdump ou tshark pour regarder ce qu'il se passe.
    Quand tu passes un appelles sortant tu as juste un blanc et puis ça fini par raccrocher ? Tu es bien enregistré et le sip options passe donc y a pas trop de raison sur ton congestion, ou alors tu balances un mauvais numéro quand tu appelles ou celui que tu appelles est bien congestion.
    Découvrez Wazo sous licence GPLv3 et accessible pour tous : http://www.wazo.community
    Blog Wazo : http://blog.wazo.community
    Wazo est un fork de XiVO.
    Suivez moi sur Twitter !

  8. #8
    Membre Association Avatar de quintana
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    Ok je viens de lire cela !

    Executing [00298337600@users:1] Dial("SIP/13-0000021d", "SIP/0298337600@ovh-out,,r")

    Tu peux changer ton dialplan et faire un Dial avec SIP/ovh-out/0298337600, ta syntaxe me paraît assez bizarre.
    Découvrez Wazo sous licence GPLv3 et accessible pour tous : http://www.wazo.community
    Blog Wazo : http://blog.wazo.community
    Wazo est un fork de XiVO.
    Suivez moi sur Twitter !

  9. #9
    Membre Junior
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    Hm j'ai du mal a comprendre ce que je dois faire.

    J'ai ça dans mes extensions:
    Code:
    [outgoing]
    exten => _0.,1,Dial(SIP/${EXTEN:1}@ovh-out,,r)
    Cette syntaxe ne serait plsu correcte pour la version 11 ? Si je comprend bien de devrait virer les ",,r" ?

  10. #10
    Membre Association Avatar de quintana
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    Code:
    [outgoing]
    exten => _0.,1,Dial(SIP/ovh-out/${EXTEN:1},,r)

    Fait comme ceci puis un dialplan reload.
    Découvrez Wazo sous licence GPLv3 et accessible pour tous : http://www.wazo.community
    Blog Wazo : http://blog.wazo.community
    Wazo est un fork de XiVO.
    Suivez moi sur Twitter !

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