distribution packagée elastix
Bonjour et bravo pour votre richesse d'échanges sur a*
Je suis nouveau et j'ai le projet de déployer un a* pour me familiariser avec l'outil.
J'ai pris un sip trunk chez IPPI.
J'ai installé une distrib elastix sur un VM avec 2 softphones en local (egika+x-lite).
La com passe bien entre les 2 SF et
mon sip show registery me donne l'info suivant:
Host dnsmgr Username Refresh State Reg.Time
ippi.fr:5060 N xxxxxxxxxxx 105 Registered Wed, 29 Jul 2015 11:00:48
mon sip show peers:
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
100/100 192.168.1.11 D No No A 5060 OK (1 ms)
200/200 192.168.1.11 D No No A 65508 OK (4 ms)
ippi-incoming 213.215.xx.xxx Yes Yes 5060 Unmonitored
ippi-out/xxxxxxxxx 213.215.xx.xxx Yes Yes 5060 Unmonitored
Pourtant, je ne peux ni émettre un appel ni en recevoir.
Pour completer, je respecte les éléments suivant:
Dans le fichier sip.conf :
include sip_registrations.conf
[ general]
...
register => identifiant_ippi:mot_de_passe_ippi@ippi.fr/04xxxxxxxx
include sip_additional.conf
...
[ippi_incoming] ; configuration des appels entrants depuis ippi
type=peer
host=ippi.fr
context=from_ippi
nat=yes
canreinvite=no
[ippi-out] ; configuration des appels sortants par ippi
type=peer
host=ippi.fr
username=identifiant_ippi
secret=mot_de_passe_ippi
fromuser=identifiant_ippi
fromdomain=ippi.fr
nat=yes
canreinvite=no
[my_phone] ; configuration d'un compte SIP sur Asterisk avec l'identifiant my_phone et le mot de passe my_password
type=friend
secret=my_password
host=dynamic
context=home
nat=yes
et dans extension.conf
include extension_additional.conf
[from-trunk-sip-ippi-out]
include => from-trunk-sip-ippi-out-custom
exten => _.,1,Set(GROUP()=OUT_1)
exten => _.,n,Goto(from-trunk,${EXTEN},1)
;--== end of [from-trunk-sip-ippi-out] ==--;
[ext-trunk]
include => ext-trunk-custom
exten => 1,1,Set(TDIAL_STRING=SIP/ippi-out)
exten => 1,n,Set(DIAL_TRUNK=1)
exten => 1,n,Goto(ext-trunk,tdial,1)
exten => tdial,1,Set(OUTBOUND_GROUP=OUT_${DIAL_TRUNK})
exten => tdial,n,GotoIf($["${OUTMAXCHANS_${DIAL_TRUNK}}" = ""]?nomax)
exten => tdial,n,GotoIf($[${GROUP_COUNT(OUT_${DIAL_TRUNK})} >= ${OUTMAXCHANS_${DIAL_TRUNK}}]?hangit)
exten => tdial,n(nomax),ExecIf($["${CALLINGPRES_SV}" != ""]?Set(CALLERPRES()=${CALLINGPRES_SV}))
exten => tdial,n,Set(DIAL_NUMBER=${FROM_DID})
exten => tdial,n,GosubIf($["${PREFIX_TRUNK_${DIAL_TRUNK}}" != ""]?sub-flp-${DIAL_TRUNK},s,1())
exten => tdial,n,Set(OUTNUM=${OUTPREFIX_${DIAL_TRUNK}}${DIA L_NUMBER})
exten => tdial,n,Set(DIAL_TRUNK_OPTIONS=${IF($["${DB_EXISTS(TRUNK/${DIAL_TRUNK}/dialopts)}" = "1"]?${DB_RESULT}:${TRUNK_OPTIONS})})
exten => tdial,n,Dial(${TDIAL_STRING}/${OUTNUM},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS })
exten => tdial,n,Set(CALLERID(number)=${CALLERID(number):0: 40})
exten => tdial,n,Set(CALLERID(name)=${CALLERID(name):0:40})
exten => tdial,n(hangit),Hangup
;--== end of [ext-trunk] ==--;
J'ai bien tenté de simplifier au max par :
Dans le fichier extension.conf :
...
[from_ippi] ; les appels entrants de ippi font sonner le compte SIP my_phone
exten => s,1,Dial(SIP/my_phone)
[ home] ; tous les appels sortants du compte my_phone sont envoyés sur le réseau ippi
exten => _X.,1,Dial(SIP/ippi_outgoing/${EXTEN})
mais rien n'y fait.
je passe à coté du problème: quelqu'un y voit-il une solution ?
Merci par avance.
ci joint le log suite a l'appel vers l'exterieure
[2015-07-29 21:50:01] VERBOSE[2564] asterisk.c: -- Remote UNIX connection
[2015-07-29 21:50:01] VERBOSE[2972] asterisk.c: -- Remote UNIX connection disconnected
[2015-07-29 21:50:44] VERBOSE[2764][C-00000001] netsock2.c: == Using SIP RTP TOS bits 184
[2015-07-29 21:50:44] VERBOSE[2764][C-00000001] netsock2.c: == Using SIP RTP CoS mark 5
[2015-07-29 21:50:44] VERBOSE[2975][C-00000001] pbx.c: -- Executing [xxxxxxxxxx@from-internal:1] ResetCDR("SIP/100-00000001", "") in new stack
[2015-07-29 21:50:44] VERBOSE[2975][C-00000001] pbx.c: -- Executing [xxxxxxxxxx@from-internal:2] NoCDR("SIP/100-00000001", "") in new stack
[2015-07-29 21:50:44] VERBOSE[2975][C-00000001] pbx.c: -- Executing [xxxxxxxxxx@from-internal:3] Progress("SIP/100-00000001", "") in new stack
[2015-07-29 21:50:44] VERBOSE[2975][C-00000001] pbx.c: -- Executing [xxxxxxxxxx@from-internal:4] Wait("SIP/100-00000001", "1") in new stack
[2015-07-29 21:50:45] VERBOSE[2975][C-00000001] pbx.c: -- Executing [xxxxxxxxxx@from-internal:5] Progress("SIP/100-00000001", "") in new stack
[2015-07-29 21:50:45] VERBOSE[2975][C-00000001] pbx.c: -- Executing [xxxxxxxxxx@from-internal:6] Playback("SIP/100-00000001", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2015-07-29 21:50:45] VERBOSE[2975][C-00000001] file.c: -- <SIP/100-00000001> Playing 'silence/1.gsm' (language 'en')
[2015-07-29 21:50:46] VERBOSE[2975][C-00000001] file.c: -- <SIP/100-00000001> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
[2015-07-29 21:50:48] VERBOSE[2975][C-00000001] file.c: -- <SIP/100-00000001> Playing 'check-number-dial-again.gsm' (language 'en')
[2015-07-29 21:50:51] VERBOSE[2975][C-00000001] pbx.c: -- Executing [xxxxxxxxxx@from-internal:7] Wait("SIP/100-00000001", "1") in new stack
[2015-07-29 21:50:52] VERBOSE[2975][C-00000001] pbx.c: -- Executing [xxxxxxxxxx@from-internal:8] Congestion("SIP/100-00000001", "20") in new stack
[2015-07-29 21:50:52] WARNING[2975][C-00000001] channel.c: Prodding channel 'SIP/100-00000001' failed
[2015-07-29 21:50:52] VERBOSE[2975][C-00000001] pbx.c: == Spawn extension (from-internal, xxxxxxxxxx, 8) exited non-zero on 'SIP/100-00000001'
[2015-07-29 21:50:52] VERBOSE[2975][C-00000001] pbx.c: -- Executing [h@from-internal:1] Hangup("SIP/100-00000001", "") in new stack
[2015-07-29 21:50:52] VERBOSE[2975][C-00000001] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000001'
