Gestion des messages vocaux d’information dans Asterisk et installation de cette Mise à jour

Oussama Hammami, 2010-07-19

Switzernet

Gestion des messages vocaux d’information dans Asterisk. 1

H323-return-code. 1

Le channel-variable ${HANGUPCAUSE} 2

Le channel-variable ${DIALSTATUS} 5

Extensions.conf 5

Personnaliser les messages vocaux d’Asterisk. 14

Enregistrement des clients sur Asterisk. 15

PortaSip. 15

Asterisk. 17

Mise en œuvre du Patch d’Astrad. 17

Installation. 17

Configurer Log Rotate pour le log du script Liste-Asterisk.pl 19

Vérification de fonctionnement du Patch : 19

Référence. 21

Gestion des messages vocaux d’information dans Asterisk

H323-return-code

Le serveur RADIUS doit utiliser le H323-return-code pour fournir la raison pour laquelle l'autorisation à été rejeté. Cette VSA (Very Small Application) doit être présent dans tous les serveurs d'authentification RADIUS y compris PortaBilling. Le tableau ci-dessous donne la liste des valeurs valides H323-return-code, leur description, le prompt à jouer et l'action à exécuter par l’Asterisk.

Return Code	Description	Prompt Played	Action	Asterisk Prompt
0	Authorization succeeded	Service dependent prompt	Continue the call	free-calls or beep
1	Invalid account	Invalid	Retry one more time	invalid_account
2	Invalid PIN	Invalid	Retry one more time	wrong_passwd
3	Account in use	Account In Use	Hang up	in_use
4	Zero balance	Zero Balance	Hang up	zero_balance
5	Card Expired	Account Expired	Hang up	account_expired
6	Credit limit in post-paid accounts	Credit Limit	Hang up	credit_limit
7	User deny	Account blocked	Hang up	user_denied
8	Service not available	Technical Problem	Hang up	not_avail
9	Destination number blocked	Destination block	Hang up	no-route-exists-to-dest
10	Number of retries exceeded	Goodbye	Hang up	retries
11	Invalid RADIUS argument	Technical Problem	Hang up	cannot-complete-temp-error
12	Insufficient balance	Insufficient Fund	Hang up	insufficient_balance
13	Toll-free call	No prompts	Continue the Call	cannot-complete-temp-error
14	Invalid card number	Invalid	Hang up	cannot-complete-temp-error
21	Invalid destination number	Invalid	Hang up	cannot-complete-temp-error
95,98,99	Database related errors	Technical Problem	Hang up	cannot-complete-temp-error
101~105	ANI related errors	Technical Problem	Hang up	cannot-complete-temp-error
111~115	Account related errors	Technical Problem	Hang up	cannot-complete-temp-error
< 0	Invalid return code	Technical Problem	Hang up	cannot-complete-temp-error
Absent	Missing return code	Technical Problem	Hang up	cannot-complete-temp-error

Le channel-variable ${HANGUPCAUSE}

Les causes de raccrochage d’Asterisk sont livrés par le dialplan dans le channel-variable ${HANGUPCAUSE} après l'exécution du la commande "dial" dans extensions.conf.

En SIP, nous avons une table de conversion qui converti les codes de retour SIP et Q.931 dans les deux sens. Il s'agit d'amélioration de la compatibilité de SIP/ISDN.

Ci dessous les codes actuels disponible dans l’Asterisk basés sur la spécification Q.850/Q.931:

ISUP Cause value		SIP response
1	unallocated number	404	Not Found
2	no route to network	404	Not found
3	no route to destination	404	Not found
16	normal call clearing	----	BYE or CANCEL
17	user busy	486	Busy here
18	no user responding	408	Request Timeout
19	no answer from the user	480	Temporarily unavailable
20	subscriber absent	480	Temporarily unavailable
21	call rejected	403	Forbidden
22	number changed (w/o diagnostic)	410	Gone
22	number changed (w/ diagnostic)	301	Moved Permanently
23	redirection to new destination	410	Gone
26	non-selected user clearing	404	Not Found
27	destination out of order	502	Bad Gateway
28	address incomplete	484	Address incomplete
29	facility rejected	501	Not implemented
31	normal unspecified	480	Temporarily unavailable
34	no circuit available	503	Service unavailable
38	network out of order	503	Service unavailable
41	temporary failure	503	Service unavailable
42	switching equipment congestion	503	Service unavailable
47	resource unavailable	503	Service unavailable
55	incoming calls barred within CUG	403	Forbidden
57	bearer capability not authorized	403	Forbidden
58	bearer capability not presently available	503	Service unavailable
65	bearer capability not implemented	488	Not Acceptable Here
70	only restricted digital avail	488	Not Acceptable Here
79	service or option not implemented	501	Not implemented
87	user not member of CUG	403	Forbidden
88	incompatible destination	503	Service unavailable
10	2 recovery of timer expiry	504	Gateway timeout
11	1 protocol error	500	Server internal error
12	7 interworking unspecified	500	Server internal error

Idem à H323, il ya des cas où un protocole d'initiation de session (SIP) d'appel peut échouer en raison d'un événement de défaillance interne.

La réponse d’Asterisk doit être analogue à la Leg IP (H.323)

Q.931 Cause Code	Description	SIP Event	Description	Asterisk Prompt
0	Uninitialized.	500	Internal server error.	----
1	Unassigned number.	410	Gone.	Not-Found
2	No route to the transit network.	500	Internal server error.	----
3	No route to the destination.	404	Not found.	Not-Found
4	Send information tone.	500	Internal server error.	----
5	Misdialed trunk prefix.	500	Internal server error.	----
6	Unacceptable channel.	500	Internal server error.	----
7	Call awarded.	500	Internal server error.	----
8	Preemption.	500	Internal server error.	----
9	Preemption reserved.	500	Internal server error.	----
16	Normal.	BYE	Bye.	----
17	Busy.	486	Busy here.	----
18	No response from the user.	480	Temporarily unavailable.	----
19	No answer from the user.	480	Temporarily unavailable.	----
20	Subscriber is absent.	500	Internal server error.	----
21	Call rejected.	603	Decline.	----
22	Number has changed.	480	Temporarily unavailable.	----
26	Selected user is clearing.	500	Internal server error.	----
27	Destination is out of order.	404	Not found.	Not-Found
28	Invalid number.	484	Address incomplete.	ISUP28
29	Facility rejected.	501	Not implemented.	----
30	Response to status inquiry.	500	Internal server error.	----
31	Normal unspecified.	404	Not found.	Not-Found
34	No circuit available.	503	Service unavailable.	all-outgoing-lines-unavailable
35	Requested VPCI VCI is not available.	500	Internal server error.	----
36	VPCI VCI assignment failure.	500	Internal server error.	----
37	Cell rate is not available.	500	Internal server error.	----
38	Network is out of order.	503	Service unavailable.	all-outgoing-lines-unavailable
39	Permanent frame mode is out of service.	500	Internal server error.	----
40	Permanent frame mode is operational.	500	Internal server error.	----
41	Temporary failure.	503	Service unavailable.	all-outgoing-lines-unavailable
42	Switch is congested.	503	Service unavailable.	all-outgoing-lines-unavailable
43	Access information has been discarded.	500	Internal server error.	----
44	No required circuit.	503	Service unavailable.	----
45	No VPCI VCI is available.	500	Internal server error.	----
46	Precedence call blocked.	500	Internal server error.	----
47	No resource available.	503	Service unavailable.	all-outgoing-lines-unavailable
48	DSP error.	500	Internal server error.	----
49	QoS is not available.	500	Internal server error.	----
50	Facility is not subscribed.	500	Internal server error.	----
53	Outgoing calls barred.	500	Internal server error.	----
55	Incoming calls barred.	603	Decline.	----
57	Bearer capability is not authorized.	501	Not implemented.	----
58	Bearer capability is not available.	501	Not implemented.	----
62	Inconsistency in the information and class.	500	Internal server error.	----
63	Service or option not available.	503	Service unavailable.	----
65	Bearer capability is not implemented.	501	Not implemented.	----
66	Change type is not implemented.	500	Internal server error.	----
69	Facility is not implemented.	500	Internal server error.	----
70	Restricted digital information only.	500	Internal server error.	----
79	Service is not implemented.	501	Not implemented.	----
81	Invalid call reference value.	500	Internal server error.	----
82	Channel does not exist.	500	Internal server error.	----
83	Call exists and call ID in use.	500	Internal server error.	----
84	Call ID in use.	500	Internal server error.	----
85	No call suspended.	500	Internal server error.	----
86	Call cleared.	500	Internal server error.	----
87	User is not in CUG.	603	Decline.	----
88	Incompatible destination.	404	Bad request.	Not-Found
90	CUG does not exist.	500	Internal server error.	cannot-complete-temp-error
91	Invalid transit network.	500	Internal server error.	cannot-complete-temp-error
93	AAL parameters not supported.	500	Internal server error.	cannot-complete-temp-error
95	Invalid message.	404	Bad request.	Not-Found
96	Mandatory Information Element (IE) is missing.	500	Internal server error.	cannot-complete-temp-error
97	Message type is not implemented.	500	Internal server error.	cannot-complete-temp-error
98	Message type is not compatible.	500	Internal server error.	cannot-complete-temp-error
99	IE is not implemented.	500	Internal server error.	cannot-complete-temp-error
100	Invalid IE contents.	500	Internal server error.	cannot-complete-temp-error
101	Message in incomplete call state.	500	Internal server error.	cannot-complete-temp-error
102	Recovery on timer expiration.	408	Request timeout.	cannot-complete-temp-error
103	Non-implemented parameter was passed on.	500	Internal server error.	cannot-complete-temp-error
110	Unrecognized parameter message discarded.	500	Internal server error.	cannot-complete-temp-error
111	Protocol error.	404	Bad request.	Not-Found
127	Internetworking error.	500	Internal server error.	cannot-complete-temp-error

Le channel-variable ${DIALSTATUS}

Contient une chaîne de texte signifiant le résultat de la dernière tentative de connexion :

DIALSTATUS	Description	Asterisk Prompt
ANSWER	en cas du succès. L'appelant a atteint l'appelé.	----
BUSY	numéro est occupé.	----
NOANSWER	Pas de réponse. L’appel a atteint son time-out de sonnerie.	----
CANCEL	appel est annulé.	cancelled
CONGESTION	congestion. Ce statut est généralement un signe que le numéro composé n'est pas reconnu	cannot-complete-otherend-error
CHANUNAVAIL	Channel indisponible.	an-error-has-occured
DONTCALL	le mode de confidentialité, l'appelé a rejeté l'appel	----
TORTURE	le mode de confidentialité, l'appelé a choisi d'envoyer l'appelant à sa liste noire	----
INVALIDARGS	Erreur d'analyse des arguments de commande Dial	----

Extensions.conf

La configuration des messages vocaux d’informations dans l’Asterisk se fait a travers le fichier ‘/etc/asterisk/extensions.conf’:

[globals]

RADIUS_Server=IP_Radius_Server(PortaBilling)

RADIUS_Secret=Password

RADIUS_Auth_Port=1812

RAIUS_Acct_Port=1813

Acct_Update_Timeout=60

NAS_IP_Address=IP_Asterisk_Server

II=0

max_duration=7200

[general]

static=yes

autofallthrough=yes

[forbidden]

exten => _X.,1,NoOp(-- No context for host ${SIPCHANINFO(recvip)} --)

exten => _X.,n,Hangup(21)

[h323-id]

exten => _X.,1,NoOp(-- H323 ID --)

exten => _X.,n,Set(ID=${SIP_HEADER(h323-conf-id)})

exten => _X.,n,ExecIf($[ "${ID}" = "" ],Set,ID=${RAND(300000000,2000000000)}-${RAND(300000000,2000000000)}-${RAND(300000000,2000000000)}-${RAND(300000000,2000000000)})

exten => _X.,n,Set(H323_ID=${ID})

exten => _X.,n,Set(H323_ID_HEX=${MATH(${CUT(ID,,1)}+0,hex)} ${MATH(${CUT(ID,,2)}+0,hex)} ${MATH(${CUT(ID,,3)}+0,hex)} ${MATH(${CUT(ID,,4)}+0,hex)})

exten => _X.,n,SIPAddHeader(h323-conf-id: ${H323_ID})

exten => _X.,n,Set(CDR(h323id)=${H323_ID_HEX})

exten => _X.,n,Return()

[auth-ip]

exten => _X.,1,NoOp(-- Inbound Authentication for cusromer IP --)

exten => _X.,n,Gosub(h323-id,${EXTEN},1)

exten => _X.,n,Set(CDR(accountcode)=${SIPCHANINFO(recvip)})

exten => _X.,n,Set(MYCALLID=${SIPCALLID})

exten => _X.,n,Set(CDR(CustomerType)=IpAuth)

exten => _X.,n,agi,agi-rad-auth-2.pl|Routing=SIP&AuthorizeBy=Account&Password=cisco&H323_ID=${H323_ID}&CustomerType=IpAuth

exten => _X.,n,Set(CDR(disconnect-reason)=${h323-return-code})

exten => _X.,n,Goto(rad-auth,${EXTEN},1)

;exten => _X.,n,Goto(routing,${EXTEN},1)

[auth-inbound]

exten => _X.,1,NoOp(-- Inbound Authentication --)

exten => _X.,n,Gosub(h323-id,${EXTEN},1)

exten => _X.,n,Set(CDR(accountcode)=${SIPCHANINFO(recvip)})

exten => _X.,n,Set(MYCALLID=${SIPCALLID})

exten => _X.,n,Set(CDR(CustomerType)=PeerAuth)

exten => _X.,n,agi,agi-rad-auth-2.pl|Routing=SIP&AuthorizeBy=Account&Password=cisco&H323_ID=${H323_ID}&CustomerType=PeerAuth

exten => _X.,n,Set(CDR(disconnect-reason)=${h323-return-code})

exten => _X.,n,Goto(rad-auth,${EXTEN},1)

;exten => _X.,n,Goto(routing,${EXTEN},1)

[sip-auth]

exten => _X.,1,NoOp(-- SIP Authentication --)

exten => _X.,n,Gosub(h323-id,${EXTEN},1)

exten => _X.,n,Set(SIP_Authorization=${SIP_HEADER(Proxy-Authorization)})

exten => _X.,n,Set(CDR(CustomerType)=SipAuth)

exten => _X.,n,agi,agi-rad-auth-2.pl|Routing=SIP&AuthorizeBy=SIP&IfFailed=DoNotHangup&H323_ID=${H323_ID}&CustomerType=SipAuth

exten => _X.,n,Set(CDR(accountcode)=${SIP_Username})

exten => _X.,n,Set(CDR(disconnect-reason)=${h323-return-code})

exten => _X.,n,Goto(rad-auth,${EXTEN},1)

le contexte [rad-auth] analyse la réponse du serveur Radius (RADIUS_Server).

Cette réponse est communiqué entre l’Asterisk et le script agi-rad-auth-2.pl a travers le variable ${h323-return-code}

[rad-auth]

exten => _X.,1,NoOp()

exten => _X.,n,GotoIf($[ ${h323-return-code} = 1 ]?invalid_account,${EXTEN},1)

exten => _X.,n,GotoIf($[ ${h323-return-code} = 2 ]?wrong_passwd,${EXTEN},1)

exten => _X.,n,GotoIf($[ ${h323-return-code} = 3 ]?in_use,${EXTEN},1)

exten => _X.,n,GotoIf($[ ${h323-return-code} = 4 ]?zero_balance,${EXTEN},1)

exten => _X.,n,GotoIf($[ ${h323-return-code} = 5 ]?account_expired,${EXTEN},1)

exten => _X.,n,GotoIf($[ ${h323-return-code} = 6 ]?credit_limit,${EXTEN},1)

exten => _X.,n,GotoIf($[ ${h323-return-code} = 7 ]?user_denied,${EXTEN},1)

exten => _X.,n,GotoIf($[ ${h323-return-code} = 8 ]?not_avail,${EXTEN},1)

exten => _X.,n,GotoIf($[ ${h323-return-code} = 9 ]?no-route-exists-to-dest,${EXTEN},1)

exten => _X.,n,GotoIf($[ ${h323-return-code} = 10 ]?retries,${EXTEN},1)

exten => _X.,n,GotoIf($[ ${h323-return-code} = 12 ]?insufficient_balance,${EXTEN},1)

exten => _X.,n,GotoIf($[ ${h323-return-code} = 0 | ${h323-return-code} = 13 ]?welcome-msg,${EXTEN},1)

exten => _X.,n,Goto(cannot-complete-temp-error,${EXTEN},1)

Dans le contexte [welcome-msg] on joue le message free-call si l’appel est gratuit (${h323-return-code} = 0) sinon une simple beep (${h323-return-code} = 13).

Pour éviter les problèmes causés par la succession du paquet SIP ‘ringing’ par le paquet ‘session progress’, on a limité ce message aux appels dont le ‘user_Agent’ est Linksys puisqu’il gère cette situation.

[welcome-msg]

exten => _X.,1,NoOp()

exten => _X.,n,GotoIf($[ ${XTOT} < 1 ]?telephone-number-temp-disconnected,${EXTEN},1)

exten => _X.,n,Set(User_Agent=${SIP_HEADER(User-Agent)})

exten => _X.,n,Set(CDR(User-Agent)=User-Agent:${User_Agent})

exten => _X.,n,GotoIf($[ "${User_Agent:0:7}" = "Linksys" ]?:nomsg)

exten => _X.,n,GotoIf($[ ${h323-return-code} = 13 ]?free-call,${EXTEN},1)

exten => _X.,n,GotoIf($[ ${h323-return-code} = 0 ]?normal-call,${EXTEN},1)

exten => _X.,n(nomsg),Goto(routing,${EXTEN},1)

[free-call]

exten => _X.,1,NoOp()

exten => _X.,n,Playback(free-calls-en,noanswer)

exten => _X.,n,Playback(free-calls-fr,noanswer)

exten => _X.,n,Goto(routing,${EXTEN},1)

[normal-call]

exten => _X.,1,NoOp()

exten => _X.,n,Playback(beep,noanswer)

exten => _X.,n,Goto(routing,${EXTEN},1)

Dans ce contexte [routing], on applique la liste de routage envoyée par le Billing via radius et analysé par le script agi-rad-auth-2.pl qui la transforme en une simulation d’une table à deux dimensions présentées par un ensemble des variables Asterisk ${XROUTE_${II}_${KK}}. http://switzernet.com/public/100427-follow-me-asterisk/

Le variable ${XTOT} est strictement supérieur à 1 seulement s’il s’agit d’un appel déviée (Follow Me est activé), dans cette cas en joue le message ‘next’ au moment de changement de la destination.

Le routage se fait à l’aide de deux boucles ‘loop’ imbriquées, en cas d’erreur on quitte le deuxième boucle ce qui implique la sortie du boucle supérieur si c’est un appel simple ou le passage à la destination suivante si c’est un appel déviée.

[routing]

exten => _X.,1,NoOp()

exten => _X.,n,Set(II=1)

exten => _X.,n(loop),NoOp()

exten => _X.,n,GotoIf($[ ${II} = 1 ]?SingleRoute)

exten => _X.,n,Playback(the-next,noanswer)

exten => _X.,n(SingleRoute),Set(KK=0)

exten => _X.,n,Set(SUM_ROUTE=${XSUM_${II}})

exten => _X.,n(loop1),NoOp()

exten => _X.,n,Set(TARGET_ROUTE=${XROUTE_${II}_${KK}})

exten => _X.,n,Set(TARGET_EXPIRE=${XEXP_${II}_${KK}})

exten => _X.,n,Set(TARGET_CLI=${XCLI_${II}_${KK}})

exten => _X.,n,Set(CALLERID(all)=${TARGET_CLI} <${TARGET_CLI}>)

exten => _X.,n,Set(MYDNID=${CUT(TARGET_ROUTE,@,1)})

exten => _X.,n,Set(VDNID=${CUT(MYDNID,/,2)})

exten => _X.,n,Dial(${IF($[ ${CUT(TARGET_ROUTE,@,2)} = ${NAS_IP_Address}]?${MYDNID}:${TARGET_ROUTE})},${TARGET_EXPIRE})

exten => _X.,n,Set(CDR(Hangupcause)=Hangupcause:${HANGUPCAUSE})

exten => _X.,n,Set(KK=$[${KK}+1])

exten => _X.,n,GotoIf($[ ${HANGUPCAUSE} = 1 | ${HANGUPCAUSE} = 3 | ${HANGUPCAUSE} = 27 | ${HANGUPCAUSE} = 28 | ${HANGUPCAUSE} = 31 | ${HANGUPCAUSE} = 88 | ${HANGUPCAUSE} >= 90 | "${DIALSTATUS}" = "BUSY" | "${DIALSTATUS}" = "NOANSWER" | "${DIALSTATUS}" = "CANCEL" ]?end1)

exten => _X.,n,GotoIf($[${KK} < ${SUM_ROUTE} ]?loop1:end1)

exten => _X.,n(end1),NoOp()

exten => _X.,n,Set(II=$[${II}+1])

exten => _X.,n,GotoIf($[${II} <= ${XTOT} ]?loop:end)

exten => _X.,n(end),NoOp()

exten => _X.,n,Goto(Error-Info,${EXTEN},1)

Dans ce contexte [Error-onfo], on joue les messages vocaux qui correspond à la valeur du retour du résultat de la dernière tentative de connexion (${HANGUPCAUSE} et ${DIALSTATUS}).

Ce contexte sera ignoré dans le cas d’un appel dévié.

[Error-Info]

exten => _X.,1,NoOp()

exten => _X.,n,GotoIf($[ ${XTOT} > 1 ]?Ter)

exten => _X.,n,GotoIf($[ ${HANGUPCAUSE} = 28 ]?ISUP28,${EXTEN},1)

exten => _X.,n,GotoIf($[ ${HANGUPCAUSE} = 1 | ${HANGUPCAUSE} = 3 | ${HANGUPCAUSE} = 27 | ${HANGUPCAUSE} = 31 | ${HANGUPCAUSE} = 88 | ${HANGUPCAUSE} = 95 | ${HANGUPCAUSE} = 111 ]?Not-Found,${EXTEN},1)

exten => _X.,n,GotoIf($[ ${HANGUPCAUSE} >= 90 ]?cannot-complete-temp-error,${EXTEN},1)

exten => _X.,n,GotoIf($[ "${DIALSTATUS}" = "CANCEL" ]?cancelled,${EXTEN},1)

exten => _X.,n,GotoIf($[ ${HANGUPCAUSE} = 34 | ${HANGUPCAUSE} = 38 | ${HANGUPCAUSE} = 41 | ${HANGUPCAUSE} = 42 | ${HANGUPCAUSE} = 47 ]?all-outgoing-lines-unavailable,${EXTEN},1)

exten => _X.,n,GotoIf($[ "${DIALSTATUS}" = "CHANUNAVAIL" ]?an-error-has-occured,${EXTEN},1)

exten => _X.,n,GotoIf($[ "${DIALSTATUS}" = "CONGESTION" ]?cannot-complete-otherend-error,${EXTEN},1)

;exten => _X.,n(Ter),SayNumber(${HANGUPCAUSE},c)

;exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n(Ter),Hangup(${HANGUPCAUSE})

[ISUP28]

exten => _X.,1,NoOp()

exten => _X.,1,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(incomplete-num-en,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(incomplete-num-fr,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(incomplete-num-en,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(incomplete-num-fr,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(incomplete-num-en,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(incomplete-num-fr,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[user_denied]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(user_denied,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[cannot-complete-otherend-error]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(cannot-complete-otherend-error,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(cannot-complete-otherend-error,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(cannot-complete-otherend-error,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[cancelled]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(cancelled,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(cancelled,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(cancelled,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[not_avail]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(not_avail,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[invalid_account]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(invalid_account,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[wrong_passwd]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(wrong_passwd,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[account_expired]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(account_expired,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[credit_limit]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(credit_limit,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[zero_balance]

exten => _X.,1,NoOp()

exten => _X.,n,wait(3)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(zero_balance,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[in_use]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(simul-call-limit-reached,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(simul-call-limit-reached,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(in_use,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(simul-call-limit-reached,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[insufficient_balance]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(insufficient_balance,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(insufficient_balance,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(insufficient_balance,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[retries]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(retries,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(retries,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(retries,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[cannot-complete-temp-error]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(cannot-complete-temp-error,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(cannot-complete-temp-error,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(cannot-complete-temp-error,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[no-route-exists-to-dest]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(no-route-exists-to-dest,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(no-route-exists-to-dest,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(no-route-exists-to-dest,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(cld_blocked,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[Not-Found]

exten => _X.,1,NoOp()

exten => _X.,n,Set(NotFoundContext=Not-Found-${IF($[ ${CUT(TARGET_ROUTE,@,2)} = ${NAS_IP_Address}]?1:2)})

exten => _X.,n,Goto(${NotFoundContext},${EXTEN},1)

[Not-Found-1]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(not-found-en,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(not-found-fr,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(not-found-en,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(not-found-fr,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(not-found-en,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(not-found-fr,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[Not-Found-2]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(you-dialed-wrong-number,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(you-dialed-wrong-number,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(you-dialed-wrong-number,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[telephone-number-temp-disconnected]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(the-number-u-dialed,noanswer)

exten => _X.,n,Playback(temp-disconnected,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(the-number-u-dialed,noanswer)

exten => _X.,n,Playback(temp-disconnected,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(the-number-u-dialed,noanswer)

exten => _X.,n,Playback(temp-disconnected,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[all-outgoing-lines-unavailable]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(all-outgoing-lines-unavailable,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(all-outgoing-lines-unavailable,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(all-outgoing-lines-unavailable,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

[an-error-has-occured]

exten => _X.,1,NoOp()

exten => _X.,n,wait(1)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(an-error-has-occured,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(an-error-has-occured,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Playback(an-error-has-occured,noanswer)

exten => _X.,n,Playback(beeperr,noanswer)

exten => _X.,n,Hangup(${HANGUPCAUSE})

Personnaliser les messages vocaux d’Asterisk

Vous pouvez créer vos propres messages vocaux d’Asterisk:

1- Enregistrer le message dans un fichier WAV (PCM sans compression),

2- Uploader ce dernier dans votre serveur Asterisk

3- Exécuter les commandes suivantes :

astrad# sox my_sound.wav -r 8000 -c 1 -s -w /root/prompt2astrad/my_sound.wav resample –ql

astrad# asterisk –r

> file convert /root/prompt2astrad/my_sound.wav wav /root/prompt2astrad/my_sound.sln

> file convert /root/prompt2astrad/my_sound.wav wav /root/prompt2astrad/my_sound.alaw

> file convert /root/prompt2astrad/my_sound.wav wav /root/prompt2astrad/my_sound.ulaw

> file convert /root/prompt2astrad/my_sound.wav wav /root/prompt2astrad/my_sound.gsm

> quit

astrad# cp my_sound.* /usr/share/asterisk/sounds/

astrad# chmod a+r /usr/share/asterisk/sounds/*

Pour écouter ou télécharger les messages vocaux cliquer ICI

Enregistrement des clients sur Asterisk

PortaSip

L’échange d’information concernant l’enregistrement des clients entre le PortaBilling et PortaSip se fait via Radius, En effet pour que ce dernier puisse jouer les messages d’erreur indiquant au client que son mot de passe est incorrecte ou que le compte utilisé est inexistant, il accepte tous les demande d’enregistrement sauf qu’il ne met pas à jour la table ‘porta-sip.location’ dans le Master (PortaBilling), sa réponse SIP peut être ‘200 Auth Failed.’ ou ‘200 Ok’ mais c’est toujours 200 ce qui veut dire autorisée.

U 85.1.188.121:52128 -> 91.999.999.999:5060

Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bKc6e6ccef565035cd1e0c90f1f09a0a17;rport.

From: "41215500329" <sip:41215500329@fr2.youroute.net>;tag=155248053.

To: "41215500329" <sip:41215500329@fr2.youroute.net>.

Call-ID: 4257033876@192_168_1_121.

CSeq: 1033 REGISTER.

Contact: "41215500329" <sip:41215500329@192.168.1.121:5060>.

Max-Forwards: 70.

User-Agent: C450 IP010720000000.

Expires: 180.

Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO.

Content-Length: 0.

U 91.999.999.999:5060 -> 85.1.188.121:52128

SIP/2.0 401 Unauthorized.

Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bKc6e6ccef565035cd1e0c90f1f09a0a17;rport=52128;received=85.1.188.121.

From: "41215500329" <sip:41215500329@fr2.youroute.net>;tag=155248053.

To: "41215500329" <sip:41215500329@fr2.youroute.net>;tag=1759cd33c30ac90f1252a7b3a6bd2b65-c940.

Call-ID: 4257033876@192_168_1_121.

CSeq: 1033 REGISTER.

WWW-Authenticate: Digest realm="fr2.youroute.net", nonce="4c46b70455103276dfcfa96403b662009c8dce2f".

Server: Sip EXpress router (0.9.6 (i386/freebsd)).

Content-Length: 0.

U 85.1.188.121:52128 -> 91.999.999.999:5060

Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK9ff87d568e0582fb249768b4b741769a;rport.

From: "41215500329" <sip:41215500329@fr2.youroute.net>;tag=155248053.

To: "41215500329" <sip:41215500329@fr2.youroute.net>.

Call-ID: 4257033876@192_168_1_121.

CSeq: 1034 REGISTER.

Contact: "41215500329" <sip:41215500329@192.168.1.121:5060>.

Authorization: Digest username="41215500329", realm="fr2.youroute.net", uri="sip:fr2.youroute.net", nonce="4c46b70455103276dfcfa96403b662009c8dce2f", response="15a0902a71c14c0e200c74ab47eddaf3".

Max-Forwards: 70.

User-Agent: C450 IP010720000000.

Expires: 180.

Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO.

Content-Length: 0.

---> RADIUS port 1812 <---

U 91.999.999.999:56008 -> 66.999.999.999:1812

.x...D......#...-.....41215500329..41215500329..

.41215500329....fr2.youroute.net.,.*4c46b70455103276dfcfa96403b662009c8dce2f....sip:fr2.youroute.net...

U 91.999.999.999:5060 -> 85.1.188.121:52128

SIP/2.0 200 OK.

Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK9ff87d568e0582fb249768b4b741769a;rport=52128;received=85.1.188.121.

From: "41215500329" <sip:41215500329@fr2.youroute.net>;tag=155248053.

To: "41215500329" <sip:41215500329@fr2.youroute.net>;tag=1759cd33c30ac90f1252a7b3a6bd2b65-e6be.

Call-ID: 4257033876@192_168_1_121.

CSeq: 1034 REGISTER.

PortaBilling: available-funds:173.06 currency:CHF.

Contact: <sip:41215500329@85.1.188.121:52128>;expires=295.

Server: Sip EXpress router (0.9.6 (i386/freebsd)).

Content-Length: 0.

---> MySQL port 3306 <---

T 91.999.999.999:61902 -> 66.999.999.999:3306 [AP]

.....insert into location (username,contact,expires,q,callid,cseq,flags,user_agent,received,i_env,domain ) values ('41215500329','sip:41215500329@85.1.188.121:52128','2010-07-21 10:59:49',-1.00 ,'4257033876@192_168_1_121',1034,1,'C450 IP010720000000',NULL,1,'91.999.999.999')

Asterisk

Pour le moment on n’utilise pas Radius pour l’enregistrement mais c’est prévu pour la prochaine version.

Dans l’Asterisk chaque client doit être déclaré dans le fichier de configuration ‘db-friends,conf’ qui été crée par le script Liste-Asterisk.pl

Ce script télécharge la liste des clients crypté depuis le serveur Web spécifié dans le variable $http_file, le décryptage se fait en se basant sur la clé indiquée dans le fichier KeyAstr qui a la même valeur que le variable $KEY dans le fichier KeyAstr.php

Afin de garder la liste à jour, ce script doit être lancé par le cron chaque 10 minute.

Pour installer cette solution vous devez télécharge 100716-php-asterisk.zip

Editer les paramètres de connexion dans le fichier connection.php

Modifier la clé de cryptage dans les deux fichiers KeyAstr et KeyAstr.php.

Uploader le répertoire ‘Web Server’ dans votre serveur Web.

Le variable $http_file_1 dans le script Liste-Asterisk.pl doit contenir le lien de script PHP uploadé, si vous avez un deuxième serveur vous pouvez l’indique dans $http_file_2

Mise en œuvre du Patch d’Astrad

Installation

Pour installer cette mise à jour d’Asterisk en commence par la création d’un répertoire de travaille qui on le nomme par exemple ‘100719-astrad-update’.

Téléchargez depuis ce dernier le ZIP qui contient les fichiers nécessaires pour ce patch :

astrad# mkdir /root/100719-astrad-update

astrad# cd /root/100719-astrad-update/

astrad:/100719-astrad-update# wget -q http://switzernet.com/2/public/100719-asterisk-update/data/100719-astrad-update.tar.gz

astrad:/100719-astrad-update# ls

100719-astrad-update.tar.gz

astrad:/100719-astrad-update# tar xzf 100719-astrad-update.tar.gz

astrad:/100719-astrad-update# ls

100719-astrad-update.tar.gz agi-rad-auth-2.pl astrad-prompt cdr_custom.conf extensions.conf KeyAstr Liste-Asterisk.pl

Editer les fichiers suivants en ajoutant les informations de connexion nécessaires :

extensions.conf

astrad:/100719-astrad-update# vi extensions.conf

[globals]

RADIUS_Server=IP_Radius_Server(PortaBilling)

RADIUS_Secret=Password

RADIUS_Auth_Port=1812

RAIUS_Acct_Port=1813

Acct_Update_Timeout=60

NAS_IP_Address=IP_Asterisk_Server

. . .

Liste-Asterisk.pl

astrad:/100719-astrad-update# vi Liste-Asterisk.pl

#! /usr/bin/perl -w

use strict;

use warnings;

use POSIX;

use Digest::MD5;

use Crypt::Rijndael;

my $http_user="Username";

my $http_passwd="Password";

my $http_file_1="http://switzernet.com/folders/100716-php-asterisk";

my $http_file_2="http://unappel.ch/folders/100716-php-asterisk";

Exécutez les commandes suivantes :

astrad:/100719-astrad-update# cp ./astrad-prompt/* /usr/share/asterisk/sounds/

astrad:/100719-astrad-update# mv /etc/asterisk/extensions.conf /etc/asterisk/old-extensions.conf

astrad:/100719-astrad-update# mv /etc/asterisk/cdr_custom.conf /etc/asterisk/old-cdr_custom.conf

astrad:/100719-astrad-update# cp cdr_custom.conf /etc/asterisk/cdr_custom.conf

astrad:/100719-astrad-update# cp extensions.conf /etc/asterisk/extensions.conf

astrad:/100719-astrad-update# mv /usr/share/asterisk/agi-bin/agi-rad-auth-2.pl /usr/share/asterisk/agi-bin/old-agi-rad-auth-2.pl

astrad:/100719-astrad-update# cp agi-rad-auth-2.pl /usr/share/asterisk/agi-bin/

astrad:/100719-astrad-update# mv /root/100105-asterisk-list/Liste-Asterisk.pl /root/100105-asterisk-list/old-Liste-Asterisk.pl

astrad:/100719-astrad-update# cp Liste-Asterisk.pl /root/100105-asterisk-list/

astrad:/100719-astrad-update# chmod +r /usr/share/asterisk/sounds/*

astrad:/100719-astrad-update# chown -R asterisk:asterisk /etc/asterisk/*.conf

astrad:/100719-astrad-update# /etc/init.d/asterisk stop ; /root/myasterisk/croncheck.sh

Configurer Log Rotate pour le log du script Liste-Asterisk.pl

Editez le fichier ‘/etc/logrotate.d/asterisk’ et ajouter les lignes suivantes à la fin :

/etc/logrotate.d/asterisk

"/root/100105-asterisk-list/AsteriskPerl.log" {

daily

rotate 15

dateext

}

Vérification de fonctionnement du Patch :

On commence par la confirmation qu’Asterisk tourne en exécutant la commande suivante, vous devez avoir la même sorti dans votre console :

astrad# ps aux | grep ast

root 3551 0.0 0.0 5608 712 pts/0 S+ 09:34 0:00 grep --color=auto ast

asterisk 4223 3.0 3.3 671016 133844 ? Ssl Jul19 27:26 /usr/sbin/asterisk -p -U asterisk

asterisk 4230 0.0 0.3 65356 12444 ? S Jul19 0:45 /usr/bin/perl /root/myasterisk/ast-rad-acc.pl

asterisk 4233 0.1 0.2 61340 10328 ? S Jul19 1:29 /usr/bin/perl /root/myasterisk/ast-reg-switz.pl

asterisk 4236 0.0 5.2 472100 210388 ? S Jul19 0:22 /usr/bin/perl /root/myasterisk/ast-resend-lost.pl

root 4732 0.0 0.0 31248 3028 ? S Jul19 0:28 /usr/bin/perl /root/myasterisk/ast-ngrep.pl

Vous devez vérifier que les fichiers extensions.conf et cdr_custom.conf appartient à l’utilisateur Asterisk et que la taille du fichier ‘db-friends.conf’ est strictement supérieur à 0 :

astrad# ls -l /etc/asterisk/

total 608

-rw-r----- 1 asterisk asterisk 336 2010-05-08 17:45 asterisk.conf

-rw-r----- 1 asterisk asterisk 107 2010-07-08 14:10 billing.conf

-rw-r--r-- 1 asterisk asterisk 90 2010-01-27 17:12 cdr.conf

-rw-r--r-- 1 asterisk asterisk 777 2010-07-15 11:47 cdr_custom.conf

-rw-r--r-- 1 asterisk asterisk 502 2010-04-19 11:25 db-customer-ip.conf

-rw-r--r-- 1 asterisk asterisk 527 2010-07-16 10:49 db-deny-node.conf

-rw-r--r-- 1 asterisk asterisk 500825 2010-07-19 18:10 db-friends.conf

-rw-r--r-- 1 asterisk asterisk 1867 2010-07-16 10:49 db-peers.conf

-rw-r--r-- 1 asterisk asterisk 15029 2010-07-15 19:40 extensions.conf

-rw-r--r-- 1 asterisk asterisk 4158 2010-03-01 15:59 extensions_old.conf

-rw-r----- 1 asterisk asterisk 402 2010-05-10 19:47 manager.conf

-rw-r----- 1 asterisk asterisk 2100 2010-05-08 15:41 modules.conf

-rw-r----- 1 asterisk asterisk 750 2010-07-15 15:01 sip.conf

Pour vérifier la configuration du CDR d’Asterisk, il suffit de chercher les détailles d’un appel de test effectué après la mise à jours dans le contenu du fichier Master.csv, on doit trouver deux nouveaux champs (Hangupcause, User-Agent)

astrad# tail -f /var/log/asterisk/cdr-custom/Master.csv

"5c5a2876 611cbf59 441a80ce 1aba594b","0900000085","02155009999","","2010-07-19 19:21:44","0","2010-07-19 19:21:35","FAILED","13","SipAuth","Hangupcause:1","User-Agent:C450 IP010720000000"

Le champ ‘Hangupcause’ apparaitra seulement en cas d’erreur (appel non effectuée)

Vérifiez le contenu du log du script ‘/root/100105-asterisk-list/Liste-Asterisk.pl’, on doit voir une date supérieur à la date de mise à jour

astrad:/etc/asterisk# tail -16 /root/100105-asterisk-list/AsteriskPerl.log

2010-07-19 19:20:01

0: Reply 486 Busy Here

0: Sleep 57 s

1: Reply 200 Ok

Total Customer skipped = 0

Total Customer decrypted = 7705

Total Error in Decryption account file = 0

md5 decrypted Accounts file is equal to md5 of existing file

md5 decrypted Node file is equal to md5 of existing file

md5 decrypted IP customers file is equal to md5 of existing file

Reloading Asterisk is not necessary

-----------------------------------------------------------------------------------------------------------------

Changer le mot de passe de votre compte depuis le Billing, essayez d’appeler un numéro avant même de changer le mot de passe dans la configuration du téléphone, vous devez écouter un message qui vous informe que le mot de passe est incorrect.

Changer le mot de passe dans la configuration du votre téléphone, l’Asterisk doit autoriser cet enregistrement après 10 minutes au maximum et la date de modification du fichier ‘/etc/asterisk/db-friends.conf’ sera changer.

Référence

http://switzernet.com/public/100427-follow-me-asterisk/

http://www.cisco.com/en/US/docs/voice_ip_comm/bts/3.5/command/reference/992M.html

http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS

http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+hangupcause

http://www.asterisk.org/doxygen/trunk/AstCauses.html

http://www.voip-info.org/wiki/view/sox

http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk

http://www.linux-tutorial.info/modules.php?name=ManPage&sec=1&manpage=sox

http://www.thegeekstuff.com/2009/05/sound-exchange-sox-15-examples-to-manipulate-audio-files/

http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk