Probleme :Your call cannot be completed as dial ''solved''

[remyuk]

salut, jobtient cet erreur quand j'esseye de faire un appel sortant.
je vous joint mes infos.

pbx*CLI> Sip show peers

Name/username              Host            Dyn Nat ACL Port     Status
101/101                    192.168.1.7      D   N   A  5940     OK (49 ms)
sipgate/1129201            217.10.79.23                5060     OK (63 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected

Voici ce que j'ai lors de mon test.

== Using SIP RTP CoS mark 5
    -- Executing [08000556688@from-internal:1] ResetCDR("SIP/101-00000004", "") in new stack
    -- Executing [08000556688@from-internal:2] NoCDR("SIP/101-00000004", "") in new stack
    -- Executing [08000556688@from-internal:3] Progress("SIP/101-00000004", "") in new stack
    -- Executing [08000556688@from-internal:4] Wait("SIP/101-00000004", "1") in new stack
    -- Executing [08000556688@from-internal:5] Playback("SIP/101-00000004", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
    -- <SIP/101-00000004> Playing 'silence/1.gsm' (language 'en')
    -- <SIP/101-00000004> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
    -- <SIP/101-00000004> Playing 'check-number-dial-again.gsm' (language 'en')
    -- Executing [08000556688@from-internal:6] Wait("SIP/101-00000004", "1") in new stack
    -- Executing [08000556688@from-internal:7] Congestion("SIP/101-00000004", "20") in new stack
  == Spawn extension (from-internal, 08000556688, 7) exited non-zero on 'SIP/101-00000004'
    -- Executing [h@from-internal:1] Macro("SIP/101-00000004", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/101-00000004", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] NoOp("SIP/101-00000004", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/101-00000004", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/101-00000004", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,10)
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/101-00000004", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,12)
    -- Executing [s@macro-hangupcall:12] Hangup("SIP/101-00000004", "") in new stack
  == Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/101-00000004' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000004'
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected

infos sip trunck

General Settings

Trunk Description:	
Outbound Caller ID: mon id	
CID Options:	
Maximum Channels:	
Disable Trunk:	Disable
Monitor Trunk Failures:	  Enable
Outgoing Dial Rules 

Dial Rules:	rien

Dial Rules Wizards:	
Outbound Dial Prefix:	rien
Outgoing Settings

Trunk Name: sipgate	
PEER Details:
disallow=all
allow=ulaw&alaw&g729&alaw&ulaw)
type=peer
fromdomain=sipgate.co.uk
host=sipgate.co.uk
insecure=very
qualify=yes
secret=xxxxx
username=xxxx


Incoming Settings



USER Context:	mon id
USER Details:
context=from-trunk
fromuser=mon id
insecure=very
secret= pw
type=user
username=mon id

Registration

Register String:
monid:pw@sipgate.co.uk/monid

Je pense que mon probleme vient du dial plan, jai fait queleque recherche mais ya rien a faire, donc vos info ou tutos sont les bien venus. see ya take care.

[remyuk]

From: "101"<sip:101@192.168.1.10>;tag=f8860cc5
To: "08000556688"<sip:08000556688@192.168.1.10>;tag=as 6e4cb0ae
Call-ID: MjMxNTgxMTViOThmNDliOTg3ZGIwMDRhNGZkMTU2MDI.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:08000556688@192.168.1.10>
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 699117025 699117025 IN IP4 192.168.1.10
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.1.10
t=0 0
m=audio 10684 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
-- Executing [08000556688@from-internal:4] Wait("SIP/101-00000007", "1") in new stack
-- Executing [08000556688@from-internal:5] Playback("SIP/101-00000007", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <SIP/101-00000007> Playing 'silence/1.gsm' (language 'en')
-- <SIP/101-00000007> Playing 'cannot-complete-as-dialed.gsm' (language 'en')

<--- SIP read from UDP:192.168.1.7:5940 --->



<------------->

<--- SIP read from UDP:192.168.1.7:5940 --->
SUBSCRIBE sip:asterisk@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5940;branch=z9hG4bK-d8754z-abecc7ceeb82160b-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:101@192.168.1.7:5940>
To: "101"<sip:101@192.168.1.10>;tag=as68828cc3
From: "101"<sip:101@192.168.1.10>;tag=17d984e5
Call-ID: NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.
CSeq: 19 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite 4 release 4.0 stamp 58832
Authorization: Digest username="101",realm="asterisk",nonce="76a7e4d8",u ri="sip:asterisk@192.168.1.10",response="7b612db84 bcef48a394e454893379f1d",algorithm=MD5
Event: message-summary
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Found peer '101' for '101' from 192.168.1.7:5940

<--- Transmitting (NAT) to 192.168.1.7:5940 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.7:5940;branch=z9hG4bK-d8754z-abecc7ceeb82160b-1---d8754z-;received=192.168.1.7;rport=5940
From: "101"<sip:101@192.168.1.10>;tag=17d984e5
To: "101"<sip:101@192.168.1.10>;tag=as68828cc3
Call-ID: NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.
CSeq: 19 SUBSCRIBE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="590d3888", stale=true
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.' in 6400 ms (Method: SUBSCRIBE)

<--- SIP read from UDP:192.168.1.7:5940 --->
SUBSCRIBE sip:asterisk@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5940;branch=z9hG4bK-d8754z-bbfcc56806814850-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:101@192.168.1.7:5940>
To: "101"<sip:101@192.168.1.10>;tag=as68828cc3
From: "101"<sip:101@192.168.1.10>;tag=17d984e5
Call-ID: NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.
CSeq: 20 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite 4 release 4.0 stamp 58832
Authorization: Digest username="101",realm="asterisk",nonce="590d3888",u ri="sip:asterisk@192.168.1.10",response="a93378aeb 1943589c56d8c8b783d64fb",algorithm=MD5
Event: message-summary
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Found peer '101' for '101' from 192.168.1.7:5940
Scheduling destruction of SIP dialog 'NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.' in 310000 ms (Method: SUBSCRIBE)

<--- Transmitting (NAT) to 192.168.1.7:5940 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:5940;branch=z9hG4bK-d8754z-bbfcc56806814850-1---d8754z-;received=192.168.1.7;rport=5940
From: "101"<sip:101@192.168.1.10>;tag=17d984e5
To: "101"<sip:101@192.168.1.10>;tag=as68828cc3
Call-ID: NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.
CSeq: 20 SUBSCRIBE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:asterisk@192.168.1.10>;expires=300
Content-Length: 0


<------------>
Reliably Transmitting (NAT) to 192.168.1.7:5940:
NOTIFY sip:101@192.168.1.7:5940 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK689d9512;rport
Max-Forwards: 70
Route: <sip:101@192.168.1.7:5940>
From: "asterisk" <sip:asterisk@192.168.1.10>;tag=as68828cc3
To: <sip:101@192.168.1.7:5940>;tag=17d984e5
Contact: <sip:asterisk@192.168.1.10>
Call-ID: NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.
CSeq: 111 NOTIFY
User-Agent: Asterisk PBX 1.6.2.13
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 92

Messages-Waiting: no
Message-Account: sip:asterisk@192.168.1.10
Voice-Message: 0/0 (0/0)

---

<--- SIP read from UDP:192.168.1.7:5940 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK689d9512;rport=506 0
Contact: <sip:101@192.168.1.7:5940>
To: <sip:101@192.168.1.7:5940>;tag=17d984e5
From: "asterisk"<sip:asterisk@192.168.1.10>;tag=as68828c c3
Call-ID: NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.
CSeq: 111 NOTIFY
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
-- <SIP/101-00000007> Playing 'check-number-dial-again.gsm' (language 'en')
-- Executing [08000556688@from-internal:6] Wait("SIP/101-00000007", "1") in new stack
-- Executing [08000556688@from-internal:7] Congestion("SIP/101-00000007", "20") in new stack

<--- Reliably Transmitting (NAT) to 192.168.1.7:5940 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.7:5940;branch=z9hG4bK-d8754z-3df9eb35f1cad59c-1---d8754z-;received=192.168.1.7;rport=5940
From: "101"<sip:101@192.168.1.10>;tag=f8860cc5
To: "08000556688"<sip:08000556688@192.168.1.10>;tag=as 6e4cb0ae
Call-ID: MjMxNTgxMTViOThmNDliOTg3ZGIwMDRhNGZkMTU2MDI.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
== Spawn extension (from-internal, 08000556688, 7) exited non-zero on 'SIP/101-00000007'
-- Executing [h@from-internal:1] Macro("SIP/101-00000007", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/101-00000007", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/101-00000007", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/101-00000007", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/101-00000007", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/101-00000007", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/101-00000007", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/101-00000007' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000007'

<--- SIP read from UDP:192.168.1.7:5940 --->
ACK sip:08000556688@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5940;branch=z9hG4bK-d8754z-3df9eb35f1cad59c-1---d8754z-;rport
Max-Forwards: 70
To: "08000556688"<sip:08000556688@192.168.1.10>;tag=as 6e4cb0ae
From: "101"<sip:101@192.168.1.10>;tag=f8860cc5
Call-ID: MjMxNTgxMTViOThmNDliOTg3ZGIwMDRhNGZkMTU2MDI.
CSeq: 2 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'MjMxNTgxMTViOThmNDliOTg3ZGIwMDRhNGZkMTU2MDI.' Method: ACK

[Jacknight]

Bonjour,
C'est du Trixbox ?
Je suppose que c'est un appel de l'extérieure vers ton 101 sur Asterisk ??? (Tu ne dis rien là dessus)

Vu que tu vois des peers avec ton "SIP Show Peers", je pense aussi que le problème vient du dial plan.

Mais cela peut aussi (et c'est souvent le cas) être un problème de contexte.

1 - Vérifie bien que le contexte "from-trunk" défini dans ton /etc/asterisk/sip.conf pour le compte correspondant à ton opérateur sipgate.co.uk existe bien dans ton dial plan /etc/asterisk/extension.conf (ou extension.ael)

2 - Vérifie que le contexte en question de ton dial plan contient bien le numéro
"monid". D'ailleurs je découvre dans le SIP show peers que monid = 1129201 ^^

...mmm... (réfléchit)

A mon avis, réflexion faite, la clé du problème réside dans cette partie

Register String:
monid:pw@sipgate.co.uk/monid

Tu vois le "monid" à la fin ?
Lorsqu'un appel entre dans ton asterisk depuis sipgate.co.uk, et bien le numéro que sipgate appele sur ton Asterisk c'est justement "monid". Autrement dit : 1129201

Donc si ton dial plan inclut ce numéro "1129201" ca fonctionnera. Si non, Asterisk te répondra tout naturellement "je trouve pas le numéro"

Autre possibilité, si par exemple, tu changes la register string comme ça :

Register String:
monid:pw@sipgate.co.uk/101

Ca devrait marcher

Ou alors

tu ajoutes le numéro 1129201 dans ton dial plan

Mais ne fait pas les deux. Soit l'un soit l'autre dac ?