problème CALLERID name [résolu]

[jeremyp3]

bonjour,
après de nombreuses recherche je post ici pour avoir votre aide

j'essaye de faire fonctionner une ligne sfr libertalk sur asterisk ça fonctionne très bien pour les appels sortant sauf que le numéro afficher est au format +33990547xxxxx ce qui n'est pas très exploitable donc pour avoir le numéro en format dison normal, selon le forum n9ws il faut ajouter dans le fichier extensions.conf la ligne :

 exten = _zX.,1,Set(CALLERID(name)=maligne)

lors ce que je passe un appel il me dit :

 SIP/2.0 603 Declined

les lignes du fichier extensions.conf concerner

exten = _9X.,1,Set(CALLERID(name)=maligne)
exten => _9X.,1,Dial(SIP/${EXTEN:1}@Neuftalk-out)

j'utilise asterisk 11.5.0

je vous remerci d'avance pour votre aide


la trace sip :

Asterisk 11.5.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 11.5.0 currently running on routerLinux (pid = 12689)
routerLinux*CLI> sip etset debug on
routerLinux*CLI> SIP Debugging enabled
routerLinux*CLI> Really destroying SIP dialog '354c19061c21f0781624f3e10e635763@127.0.0.1' Method: REGISTER
routerLinux*CLI> Really destroying SIP dialog '253afee61dc56cfa2b38ca20409f8d10@127.0.0.1' Method: REGISTER
routerLinux*CLI> Really destroying SIP dialog '05281a0c778fefe47331fb1803a61d88@127.0.0.1' Method: REGISTER
routerLinux*CLI> 
<--- SIP read from UDP:10.0.2.4:6050 --->
INVITE sip:90612345678@xxx.xxx.xxx.xxx:6050 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.4:6050;branch=z9hG4bK-17535bb3
From: <sip:jeremyp31987@xxx.xxx.xxx.xxx>;tag=e51ad75840cea8c7o0
To: <sip:90612345678@xxx.xxx.xxx.xxx>
Remote-Party-ID: <sip:jeremyp31987@xxx.xxx.xxx.xxx:6050>;screen=yes;party=calling
Call-ID: a48f136c-e8ef7bc3@10.0.2.4
CSeq: 101 INVITE
Max-Forwards: 70
Contact: <sip:jeremyp31987@10.0.2.4:6050>
Expires: 240
User-Agent: Linksys/SPA2102-5.2.13(004)
Content-Length: 434
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 140736 140736 IN IP4 10.0.2.4
s=-
c=IN IP4 10.0.2.4
t=0 0
m=audio 17402 RTP/AVP 8 0 2 4 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (15 headers 20 lines) ---
Sending to 10.0.2.4:6050 (no NAT)
Sending to 10.0.2.4:6050 (no NAT)
Using INVITE request as basis request - a48f136c-e8ef7bc3@10.0.2.4
Found peer 'jeremyp31987' for 'jeremyp31987' from 10.0.2.4:6050
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G723 for ID 4
Found audio description format G729a for ID 18
Found unknown media description format G726-40 for ID 96
Found unknown media description format G726-24 for ID 97
Found unknown media description format G726-16 for ID 98
Found unknown media description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(g723|ulaw|alaw|g726|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.2.4:17402
Looking for 90612345678 in appel-sortant (domain xxx.xxx.xxx.xxx)
list_route: hop: <sip:jeremyp31987@10.0.2.4:6050>

<--- Transmitting (NAT) to 10.0.2.4:6050 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.2.4:6050;branch=z9hG4bK-17535bb3;received=10.0.2.4;rport=6050
From: <sip:jeremyp31987@xxx.xxx.xxx.xxx>;tag=e51ad75840cea8c7o0
To: <sip:90612345678@xxx.xxx.xxx.xxx>
Call-ID: a48f136c-e8ef7bc3@10.0.2.4
CSeq: 101 INVITE
Server: Asterisk PBX 11.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:90612345678@10.0.2.20:6050>
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'a48f136c-e8ef7bc3@10.0.2.4' in 6400 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 10.0.2.4:6050 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 10.0.2.4:6050;branch=z9hG4bK-17535bb3;received=10.0.2.4;rport=6050
From: <sip:jeremyp31987@xxx.xxx.xxx.xxx>;tag=e51ad75840cea8c7o0
To: <sip:90612345678@xxx.xxx.xxx.xxx>;tag=as1c5b0236
Call-ID: a48f136c-e8ef7bc3@10.0.2.4
CSeq: 101 INVITE
Server: Asterisk PBX 11.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

jerem