fichier call ne fonctionne pas

[jeremyb]

Encore merci jean pour m'avoir répondu hier , alors après avoir lancer le debug d'asterisk, j'ai fait un mv vers outgoing et voila les messages d'erreurs :

localhost*CLI> pri set debug file [test.call]
PRI debug output will be sent to '[test.call]'
[2014-07-03 09:06:52] WARNING[20486]: pbx_spool.c:285 apply_outgoing: Unknown keyword 'Context ' at line 5 of /var/spool/asterisk/outgoing/test.call
[2014-07-03 09:06:52] WARNING[20486]: pbx_spool.c:285 apply_outgoing: Unknown keyword 'Extension ' at line 6 of /var/spool/asterisk/outgoing/test.call
[2014-07-03 09:06:52] WARNING[20486]: pbx_spool.c:289 apply_outgoing: At least one of app or extension must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/test.call
[2014-07-03 09:06:52] WARNING[20486]: pbx_spool.c:456 scan_service: Invalid file contents in /var/spool/asterisk/outgoing/test.call, deleting
[2014-07-03 09:07:40] WARNING[20467]: chan_sip.c:23507 handle_response_register: Got 423 Interval too brief for service 0980487667@freephonie.net, minimum is 1800 seconds
-- Registered SIP '1002' at 192.168.1.54:34566
[2014-07-03 09:07:57] NOTICE[20467]: chan_sip.c:23645 handle_response_peerpoke: Peer '1002' is now Reachable. (9ms / 2000ms)

quand je change @192.168.1.80 par @freephonie.net voila ce qu'il se passe :

PRI debug output will be sent to '[test.call]'
[2014-07-03 09:23:55] WARNING[20486]: pbx_spool.c:285 apply_outgoing: Unknown keyword 'Context ' at line 5 of /var/spool/asterisk/outgoing/test.call
[2014-07-03 09:23:55] WARNING[20486]: pbx_spool.c:285 apply_outgoing: Unknown keyword 'Extension ' at line 6 of /var/spool/asterisk/outgoing/test.call
[2014-07-03 09:23:55] WARNING[20486]: pbx_spool.c:289 apply_outgoing: At least one of app or extension must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/test.call
[2014-07-03 09:23:55] WARNING[20486]: pbx_spool.c:456 scan_service: Invalid file contents in /var/spool/asterisk/outgoing/test.call, deleting

les messages d'erreurs sont les mêmes lorsque je mets seulement "SIP/00240XXXXXX"

Je ne vois pas trop à quoi ca correspond mais j'espère que ca t'aidera

[jean]

[2014-07-03 09:06:52] WARNING[20486]: pbx_spool.c:285 apply_outgoing: Unknown keyword 'Context ' at line 5 of /var/spool/asterisk/outgoing/test.call
[2014-07-03 09:06:52] WARNING[20486]: pbx_spool.c:285 apply_outgoing: Unknown keyword 'Extension ' at line 6 of /var/spool/asterisk/outgoing/test.call
[2014-07-03 09:06:52] WARNING[20486]: pbx_spool.c:289 apply_outgoing: At least one of app or extension must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/test.call
[2014-07-03 09:06:52] WARNING[20486]: pbx_spool.c:456 scan_service: Invalid file contents in /var/spool/asterisk/outgoing/test.call, deleting

ben j'crois qu'c'est claire, serge.... https://www.youtube.com/watch?v=u99WjU47-Sc

Il ne reconnait pas tes mots clés... Context, Extension => je pense que l'espace entre le mot clé et le : est à supprimer
channel: essaie Channel: SIP/freephonie/00240xxx

[jeremyb]

merci, alors j'ai essayé comme tu m'as dis et voila le résultat :

localhost*CLI> pri set debug file [test_v3.call]
PRI debug output will be sent to '[test_v3.call]'
[2014-07-03 15:40:44] WARNING[20486]: pbx_spool.c:309 safe_append: Unable to setutime on /var/spool/asterisk/outgoing/test_v3.call: Operation not permitted
-- Attempting call on SIP/freephonie/00240160144 for s@test:1 (Retry 1)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
[2014-07-03 15:40:44] WARNING[20451]: res_odbc.c:1412 _ast_odbc_request_obj2: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 5.1 Driver]MySQL server has gone away
[2014-07-03 15:40:44] WARNING[20451]: res_odbc.c:646 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 5.1 Driver][mysqld-5.1.73]MySQL server has gone away (65)
[2014-07-03 15:40:44] WARNING[20451]: res_odbc.c:658 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to asteriskcdrdb [MySQL-asteriskcdrdb]...
[2014-07-03 15:40:44] WARNING[20451]: res_odbc.c:762 ast_odbc_sanity_check: Conn ection is down attempting to reconnect...
[2014-07-03 15:40:44] NOTICE[20451]: res_odbc.c:1537 odbc_obj_connect: Connecting asteriskcdrdb
[2014-07-03 15:40:44] NOTICE[20451]: res_odbc.c:1569 odbc_obj_connect: res_odbc:Connected to asteriskcdrdb [MySQL-asteriskcdrdb]
-- Got SIP response 500 "Internal Server Error" back from 212.27.52.5:5060
> Channel SIP/freephonie-00000010 was never answered.
[2014-07-03 15:40:54] NOTICE[966]: pbx_spool.c:389 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy)
[2014-07-03 15:40:54] WARNING[966]: pbx_spool.c:309 safe_append: Unable to setutime on /var/spool/asterisk/outgoing/test_v3.call: Operation not permitted

En enlevant freephonie on a :

PRI debug output will be sent to '[test_v4.call]'
[2014-07-03 15:45:35] WARNING[20486]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/test_v4.call: Operation not permitted
-- Attempting call on SIP/00240160144 for s@test:1 (Retry 1)
[2014-07-03 15:45:35] WARNING[974][C-0000000f]: chan_sip.c:6163 create_addr: Purely numeric hostname (00240160144), and not a peer--rejecting!
[2014-07-03 15:45:35] NOTICE[974][C-0000000f]: channel.c:5661 __ast_request_and_dial: Unable to request channel SIP/00240160144
[2014-07-03 15:45:35] NOTICE[974]: pbx_spool.c:389 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?)
[2014-07-03 15:45:35] WARNING[974]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/test_v4.call: Operation not permitted

et mon test_v4.call ne se supprime pas

en mettant @192... on a :

PRI debug output will be sent to '[test_v5.call]'
[2014-07-03 15:51:47] WARNING[20486]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/test_v5.call: Operation not permitted
-- Attempting call on SIP/00240160144@192.168.1.80 for s@test:1 (Retry 1)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [00240160144@from-sip-external:1] NoOp("SIP/192.168.1.80-00000014", "Received incoming SIP connection from unknown peer to 00240160144") in new stack
-- Executing [00240160144@from-sip-external:2] Set("SIP/192.168.1.80-00000014", "DID=00240160144") in new stack
-- Executing [00240160144@from-sip-external:3] Goto("SIP/192.168.1.80-00000014", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/192.168.1.80-00000014", "0?checklang:noanonymous") in new stack
-- Goto (from-sip-external,s,5)
-- Executing [s@from-sip-external:5] Set("SIP/192.168.1.80-00000014", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2014-07-03 15:52:02.941 CEST.
-- Executing [s@from-sip-external:6] Answer("SIP/192.168.1.80-00000014", "") in new stack
> Channel SIP/192.168.1.80-00000013 was answered
[2014-07-03 15:51:47] WARNING[989][C-00000012]: pbx.c:6640 __ast_pbx_run: Channel 'SIP/192.168.1.80-00000013' sent to invalid extension but no invalid handler: context,exten,priority=test,s,1
== Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/192.168.1.80-00000014'
-- Executing [h@from-sip-external:1] Hangup("SIP/192.168.1.80-00000014", "") in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/192.168.1.80-00000014'
[2014-07-03 15:51:47] NOTICE[989]: pbx_spool.c:402 attempt_thread: Call completed to SIP/00240160144@192.168.1.80

je suis largué ^^'

[jean]

ca vire au gloubiboulga....

pri set debug file [test_v3.call]
=> je connais pas cette commande, mais j'ai l'impression que tu lances un debug dans un fichier qui a le meme nom que le .call - si tu peux la virer

ensuite, c'est clairement la version 4 la meilleure, on cause avec free:
-- Attempting call on SIP/freephonie/00240160144 for s@test:1 (Retry 1)
[.............]
-- Got SIP response 500 "Internal Server Error" back from 212.27.52.5:5060
> Channel SIP/freephonie-00000010 was never answered.

ok - il refuse, mais on lui parle - je pense que le numéro est incorrect

ensuite, j'ai l'impression que la partie asterisk realtime / mysql part en sucette.... peux tu faire un appel simplement (un tel/softphone connecté qui fait un dial sur le trunk freephonie) ?

[jeremyb]

Capture.JPGalors je ne savais pas quelle commande utiliser pour voir le trafic au moment de l'envoi du fichier a outgoing du coup j'utilise la commande pri set debug file[] qui va créer fichier du meme nom que le fichier observé mais avec des crochets (voir capture en piecejointe)

Sinon en connectant un softphone (xlite) j'arrive a appelé mon fixe en tappant 00240XXXXXX donc je suppose que c'est ca que je dois mettre dans mon fichier call pour faire l'appel

oublié la pièce jointe ^^'

[jeremyb]

d'ailleurs bizarrement j'ai utilisé la meme config pour 4-5 softphone différent (avec freephonie) et un seul arrive a passer des appels

[jeremyb]

je viens de voir, mon serveur avance d'un mois non ? ca risque pas de poser des problèmes ?