Bonjour Reaper,
La borne est un Siemens C590IP.
Dans les paramètres VoIP avancés, configurés en RFC 2833 (seule case cochée), le champ "Utilisez la touche R pour initier le transfert d'appel avec le protocole SIP" est à non et "Transférer l'appel en raccrochant" à oui.
L'interphone est sur le port FXO DAHDI/2-1 et la réponse se fait sur le port FXS DAHDI/3-1.
Dans le CLI, j'ai les messages suivants (le premier appui sur R se fait à 14:00:49 et j'obtiens la tonalité d'attente, le second appui sur R se fait à 14:00:51 et je récupère le correspondant sur l'interphone) :
Donc, il semble bien que le flash envoyé par la borne est intercepté et traité par la fonction REFER d'Asterisk.Code:[2012-07-30 14:00:29] VERBOSE[9894] app_dial.c: -- DAHDI/3-1 is ringing [2012-07-30 14:00:31] VERBOSE[9894] app_dial.c: -- DAHDI/3-1 answered DAHDI/2-1 [2012-07-30 14:00:31] VERBOSE[9894] pbx.c: -- Executing [s@macro-auto-blkvm:1] Set("DAHDI/3-1", "__MACRO_RESULT=") in new stack [2012-07-30 14:00:31] VERBOSE[9894] pbx.c: -- Executing [s@macro-auto-blkvm:2] Macro("DAHDI/3-1", "blkvm-clr,") in new stack [2012-07-30 14:00:31] VERBOSE[9894] pbx.c: -- Executing [s@macro-blkvm-clr:1] Set("DAHDI/3-1", "SHARED(BLKVM,DAHDI/2-1)=") in new stack [2012-07-30 14:00:31] VERBOSE[9894] pbx.c: -- Executing [s@macro-blkvm-clr:2] Set("DAHDI/3-1", "GOSUB_RETVAL=") in new stack [2012-07-30 14:00:31] VERBOSE[9894] pbx.c: -- Executing [s@macro-blkvm-clr:3] MacroExit("DAHDI/3-1", "") in new stack [2012-07-30 14:00:31] VERBOSE[9894] pbx.c: -- Executing [s@macro-auto-blkvm:3] ExecIf("DAHDI/3-1", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=3)") in new stack [2012-07-30 14:00:31] VERBOSE[9894] pbx.c: -- Executing [s@macro-auto-blkvm:4] ExecIf("DAHDI/3-1", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=)") in new stack [2012-07-30 14:00:49] VERBOSE[9894] sig_analog.c: -- Started three way call on channel 3 [2012-07-30 14:00:49] VERBOSE[9894] res_musiconhold.c: -- Started music on hold, class 'default', on DAHDI/2-1 [2012-07-30 14:00:49] VERBOSE[9905] sig_analog.c: -- Starting simple switch on 'DAHDI/3-2' [2012-07-30 14:00:51] VERBOSE[9905] sig_analog.c: -- Dumping incomplete call on DAHDI/3-1 [2012-07-30 14:00:51] VERBOSE[9894] res_musiconhold.c: -- Stopped music on hold on DAHDI/2-1 [2012-07-30 14:00:51] VERBOSE[9905] sig_analog.c: -- Hanging up on 'DAHDI/3-2' [2012-07-30 14:00:51] VERBOSE[9905] chan_dahdi.c: -- Hungup 'DAHDI/3-2' [2012-07-30 14:01:02] VERBOSE[9894] pbx.c: -- Executing [h@macro-dial:1] Macro("DAHDI/2-1", "hangupcall") in new stack [2012-07-30 14:01:02] VERBOSE[9894] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("DAHDI/2-1", "1?theend") in new stack [2012-07-30 14:01:02] VERBOSE[9894] pbx.c: -- Goto (macro-hangupcall,s,3) [2012-07-30 14:01:02] VERBOSE[9894] pbx.c: -- Executing [s@macro-hangupcall:3] Hangup("DAHDI/2-1", "") in new stack [2012-07-30 14:01:02] VERBOSE[9894] app_macro.c: == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'DAHDI/2-1' in macro 'hangupcall' [2012-07-30 14:01:02] VERBOSE[9894] features.c: == Spawn extension (macro-dial, h, 1) exited non-zero on 'DAHDI/2-1' [2012-07-30 14:01:02] VERBOSE[9894] sig_analog.c: -- Hanging up on 'DAHDI/3-1' [2012-07-30 14:01:02] VERBOSE[9894] chan_dahdi.c: -- Hungup 'DAHDI/3-1' [2012-07-30 14:01:02] VERBOSE[9894] app_macro.c: == Spawn extension (macro-dial, s, 38) exited non-zero on 'DAHDI/2-1' in macro 'dial' [2012-07-30 14:01:02] VERBOSE[9894] pbx.c: == Spawn extension (ext-group, 602, 13) exited non-zero on 'DAHDI/2-1' [2012-07-30 14:01:02] VERBOSE[9894] sig_analog.c: -- Hanging up on 'DAHDI/2-1' [2012-07-30 14:01:02] VERBOSE[9894] chan_dahdi.c: -- Hungup 'DAHDI/2-1'
Si je reçois l'appel sur une extension SIP et non DAHDI, le comportement est différent. Il me demande de numéroter une extension pour initier un transfert :
Mon problème reste entier. J'ai essayé de nombreuses combinaisons sur la borne C590 IP mais il semble qu'Asterisk intercepte systématiquement la touche R.Code:[2012-07-30 17:31:14] VERBOSE[13025] app_dial.c: -- SIP/33-0000009c is ringing [2012-07-30 17:31:16] VERBOSE[13025] app_dial.c: -- SIP/33-0000009c answered DAHDI/2-1 [2012-07-30 17:31:16] VERBOSE[13025] pbx.c: -- Executing [s@macro-auto-blkvm:1] Set("SIP/33-0000009c", "__MACRO_RESULT=") in new stack [2012-07-30 17:31:16] VERBOSE[13025] pbx.c: -- Executing [s@macro-auto-blkvm:2] Macro("SIP/33-0000009c", "blkvm-clr,") in new stack [2012-07-30 17:31:16] VERBOSE[13025] pbx.c: -- Executing [s@macro-blkvm-clr:1] Set("SIP/33-0000009c", "SHARED(BLKVM,DAHDI/2-1)=") in new stack [2012-07-30 17:31:16] VERBOSE[13025] pbx.c: -- Executing [s@macro-blkvm-clr:2] Set("SIP/33-0000009c", "GOSUB_RETVAL=") in new stack [2012-07-30 17:31:16] VERBOSE[13025] pbx.c: -- Executing [s@macro-blkvm-clr:3] MacroExit("SIP/33-0000009c", "") in new stack [2012-07-30 17:31:16] VERBOSE[13025] pbx.c: -- Executing [s@macro-auto-blkvm:3] ExecIf("SIP/33-0000009c", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=33)") in new stack [2012-07-30 17:31:16] VERBOSE[13025] pbx.c: -- Executing [s@macro-auto-blkvm:4] ExecIf("SIP/33-0000009c", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Personnel)") in new stack [2012-07-30 17:31:26] NOTICE[13025] res_rtp_asterisk.c: Unknown RTP codec 127 received from '192.168.0.252:5048' [2012-07-30 17:31:26] NOTICE[13025] res_rtp_asterisk.c: Unknown RTP codec 127 received from '192.168.0.252:5048' [2012-07-30 17:31:26] NOTICE[13025] res_rtp_asterisk.c: Unknown RTP codec 127 received from '192.168.0.252:5048' [2012-07-30 17:31:26] NOTICE[13025] res_rtp_asterisk.c: Unknown RTP codec 127 received from '192.168.0.252:5048' [2012-07-30 17:31:26] NOTICE[13025] res_rtp_asterisk.c: Unknown RTP codec 127 received from '192.168.0.252:5048' [2012-07-30 17:31:26] VERBOSE[13025] res_musiconhold.c: -- Started music on hold, class 'default', on DAHDI/2-1 [2012-07-30 17:31:26] VERBOSE[13025] file.c: -- <SIP/33-0000009c> Playing 'pbx-transfer.gsm' (language 'fr') [2012-07-30 17:31:26] NOTICE[13025] res_rtp_asterisk.c: Unknown RTP codec 127 received from '192.168.0.252:5048' [2012-07-30 17:31:26] NOTICE[13025] res_rtp_asterisk.c: Unknown RTP codec 127 received from '192.168.0.252:5048' [2012-07-30 17:31:30] WARNING[13025] features.c: No digits dialed for atxfer. [2012-07-30 17:31:30] VERBOSE[13025] file.c: -- <SIP/33-0000009c> Playing 'pbx-invalid.gsm' (language 'fr') [2012-07-30 17:31:34] VERBOSE[13025] res_musiconhold.c: -- Stopped music on hold on DAHDI/2-1
Comment faire pour changer ce comportement sur l'interface web de la borne ?