Call from 'client-3' to extension '5' rejected because extension not found in context 'users'.

[sunsetlive]

Bonjour à tous !

Dans le cadre d'un projet, je dois monter une maquette en local avec des clients X-Lite (client-1 & client-2) et un téléphone CISCO 7942 (client-3). Le problème c'est que la communication s'effectue très bien entre les clients X-LITE mais du téléphone CISCO, je
peux uniquement recevoir des appels mais impossible d'émettre un appel depuis le téléphone.

Dans les logs Asterisk, j'obtiens ceci:

NOTICE[10163]: chan_sip.c:22718 handle_request_invite: Call from 'client-3' (192.168.1.158:49769) to extension '5' rejected because extension not found in context 'users'.
lol*CLI>

fichier SIP.conf

[client-1] ; nom du téléphone
type=friend ; type de téléphone
host=dynamic
context=users
permit=192.168.1.254 ; network setting
secret=serveur
callerid="client-1" <555> ; association user et num de tel
mailbox=client-1@192.168.1.254


[client-2]
type=friend ; type de téléphone
host=dynamic
context=users
permit=192.168.1.254 ; network setting
username=client-2 ; nom d'utilisateur associé
secret=serveur ; mot de passe
callerid="client-2" <556> ; association user et num de tel
mailbox=client-2@192.168.1.254 ;Adresse de la boite vocale et dans
;notre cas remplacer nomdomaine par
;l’adresse ip de serveur asterisk

[client-3] ; nom du téléphone
type=friend ; type de téléphone
host=dynamic ; enregistrement dynamique de
context=users
permit=192.168.1.254 ; network setting
qualify = no
nat = no
allowguest = no
username=client-3 ; nom d'utilisateur associé
secret=serveur ; mot de passe
callerid="client-3" <557> ; association user et num de tel
mailbox=client-3@192.168.1.254 ;Adresse de la boite vocale et dans

extensions.conf

[general]
static=yes
writeprotect=no
autofallthrough=yes

[users]
exten => 555,1,Dial(SIP/client-1,20)
exten => 556,1,Dial(SIP/client-2,20)
exten => 557,1,Dial(SIP/client-3,20)


MERCI d'avance ! pour ceux qui m'aide !!

[jean]

tu numérotes "5" et cela n'existe pas dans ton contexte users - ce qui est effectivement le cas

Essaie de numeroter 556 ou 555...

[sunsetlive]

Merci Jean d'avoir répondu !

Non effectivement j'ai peut-être omis de dire que j'ai effectué plusieurs test !! Notamment la numérotation "555" et "556" pour joindre les client 1 & 2. Mais rien à faire lorsque je numérote que ce soit un nombre à 3 chiffres "435" ou 10 chiffres "2535454531" dans les logs de Asterisk il sort toujours "... to extension '5' ... alors qu'il devrait correspondre à la numérotation effectué sur l'IPphone. Par exemple si je compose '555' sur l'IPphone, il devrait me sortir en log:

Call from 'client-3' to extension '555' rejected because extension not found in context 'users'.

Hors C'est pas le cas !!! Il sort toujours une erreur de type Call from 'client-3' to extension '5' rejected because extension not found in context 'users' --> Un seul chiffre alors que j'en est composé 3 !!

Je sais pas si je suis claire dans mes propos mais voilà le souci ...

Merci encore d'avoir répondu !!

[jean]

au passage: c'est excellent d'avoir mis allowguest=no, mais il vaut mieux le mettre dans la section [generral]

fais un core set verbose 9
puis sip set debug on
et copie le résultat

J

[sunsetlive]

J'ai fais comme tu m'a dis, j'ai mis allowguest = no dans general
puis tapez les commandes ...
J'espère que c'est les logs attendu ...

<--- Reliably Transmitting (no NAT) to 192.168.1.158:5061 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bKc2be2d2a;received =192.168.1.158
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a5620017ccbc8736-cc8044ca
To: <sip:5@192.168.1.254>;tag=as31595924
Call-ID: a40cc394-a5620011-2b015b06-6de2c5da@192.168.1.158
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Jul 2 17:44:40] NOTICE[12769]: chan_sip.c:22718 handle_request_invite: Call from 'client-3' (192.168.1.158:52726) to extension '5' rejected because extension not found in context 'users'.
Scheduling destruction of SIP dialog 'a40cc394-a5620011-2b015b06-6de2c5da@192.168.1.158' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.158:51110 --->
ACK sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bKc2be2d2a
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a5620017ccbc8736-cc8044ca
To: <sip:5@192.168.1.254>;tag=as31595924
Call-ID: a40cc394-a5620011-2b015b06-6de2c5da@192.168.1.158
Max-Forwards: 70
Date: Tue, 02 Jul 2013 15:44:37 GMT
CSeq: 102 ACK
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'a40cc394-a5620011-2b015b06-6de2c5da@192.168.1.158' Method: ACK

[jean]

il manque des infos - fais d'abord le sip set debug on, puis lance l'appel

[sunsetlive]

J'ai effectué la commande sip set debug on puis l'appel du client-3 mais j'obtiens toujours ce genre de logs ...


<--- SIP read from UDP:192.168.1.158:50296 --->
BYE sip:555@192.168.1.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bK54886b8e
From: <sip:557@192.168.1.158:5061;transport=udp>;tag=a40 cc394a562001c21929f90-1537ed53
To: "client-1" <sip:555@192.168.1.254>;tag=as6c2b921b
Call-ID: 2db2795f36d6bb4733362aad3d972fe6@192.168.1.254:506 0
Max-Forwards: 70
Date: Wed, 03 Jul 2013 06:52:08 GMT
CSeq: 101 BYE
User-Agent: Cisco-CP7942G/9.3.1
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.1.158:5061 (no NAT)
Scheduling destruction of SIP dialog '2db2795f36d6bb4733362aad3d972fe6@192.168.1.254:50 60' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.1.158:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bK54886b8e;received =192.168.1.158
From: <sip:557@192.168.1.158:5061;transport=udp>;tag=a40 cc394a562001c21929f90-1537ed53
To: "client-1" <sip:555@192.168.1.254>;tag=as6c2b921b
Call-ID: 2db2795f36d6bb4733362aad3d972fe6@192.168.1.254:506 0
CSeq: 101 BYE
Server: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
set_destination: Parsing <sip:client-1@192.168.1.150:5060> for address/port to send to
set_destination: set destination to 192.168.1.150:5060
Audio is at 11478
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.150:5060:
INVITE sip:client-1@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK0bfe3f18;rport
Max-Forwards: 70
From: <sip:557@192.168.1.254:5060>;tag=as16824010
To: "client-1"<sip:client-1@192.168.1.254:5060>;tag=bb875e41
Contact: <sip:557@192.168.1.254:5060>
Call-ID: YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 781600293 781600295 IN IP4 192.168.1.254
s=Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
c=IN IP4 192.168.1.254
t=0 0
m=audio 11478 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.1.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK0bfe3f18;rport=50 60
Contact: <sip:client-1@192.168.1.150:5060>
To: "client-1"<sip:client-1@192.168.1.254:5060>;tag=bb875e41
From: <sip:557@192.168.1.254:5060>;tag=as16824010
Call-ID: YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.0.0 stamp 67284
Content-Length: 213

v=0
o=- 13014543165961468 3 IN IP4 192.168.1.150
s=CounterPath X-Lite 5.0.0
c=IN IP4 192.168.1.150
b=AS:1638
t=0 0
m=audio 5062 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (12 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.150:5062

[sunsetlive]

suite des logs

set_destination: Parsing <sip:client-1@192.168.1.150:5060> for address/port to send to
set_destination: set destination to 192.168.1.150:5060
Transmitting (NAT) to 192.168.1.150:5060:
ACK sip:client-1@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK767cf41d;rport
Max-Forwards: 70
From: <sip:557@192.168.1.254:5060>;tag=as16824010
To: "client-1"<sip:client-1@192.168.1.254:5060>;tag=bb875e41
Contact: <sip:557@192.168.1.254:5060>
Call-ID: YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Content-Length: 0


---
== Spawn extension (users, 557, 1) exited non-zero on 'SIP/client-1-00000000'
Scheduling destruction of SIP dialog 'YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:client-1@192.168.1.150:5060> for address/port to send to
set_destination: set destination to 192.168.1.150:5060
Reliably Transmitting (NAT) to 192.168.1.150:5060:
BYE sip:client-1@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK42bf98f8;rport
Max-Forwards: 70
From: <sip:557@192.168.1.254:5060>;tag=as16824010
To: "client-1"<sip:client-1@192.168.1.254:5060>;tag=bb875e41
Call-ID: YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Proxy-Authorization: Digest username="client-1", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.254", nonce="", response="323d061907b0821b1decc513156e9f67"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.150:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK42bf98f8;rport=50 60
Contact: <sip:client-1@192.168.1.150:5060>
To: "client-1"<sip:client-1@192.168.1.254:5060>;tag=bb875e41
From: <sip:557@192.168.1.254:5060>;tag=as16824010
Call-ID: YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.
CSeq: 104 BYE
User-Agent: X-Lite release 5.0.0 stamp 67284
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'YTg0N2E4NDI3OWEzN2IzOTQ0YmI0YmI0OTRiN2M1ZGY.' Method: ACK

<--- SIP read from UDP:192.168.1.150:5060 --->


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

<--- SIP read from UDP:192.168.1.158:50296 --->
INVITE sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bK386bb6e8
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Max-Forwards: 70
Date: Wed, 03 Jul 2013 06:52:20 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7942G/9.3.1
Contact: <sip:557@192.168.1.158:5061;transport=udp>
Expires: 180
Accept: application/sdp
llow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTE R,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 354
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 12876 0 IN IP4 192.168.1.158
s=SIP Call
t=0 0
m=audio 23956 RTP/AVP 0 8 18 102 116 101
c=IN IP4 192.168.1.158
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (18 headers 16 lines) ---
Sending to 192.168.1.158:50296 (NAT)
Using INVITE request as basis request - a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Found peer 'client-3' for 'client-3' from 192.168.1.158:50296

<--- Reliably Transmitting (no NAT) to 192.168.1.158:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bK386bb6e8;received =192.168.1.158
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>;tag=as7134625c
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="02702734"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.158:53122 --->
ACK sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bK386bb6e8
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>;tag=as7134625c
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Max-Forwards: 70
Date: Wed, 03 Jul 2013 06:52:20 GMT
CSeq: 101 ACK
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.158:50296 --->
INVITE sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bKfc3dd1e1
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Max-Forwards: 70
Date: Wed, 03 Jul 2013 06:52:20 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7942G/9.3.1
Contact: <sip:557@192.168.1.158:5061;transport=udp>
Authorization: Digest username="client-3",realm="asterisk",uri="sip:5@192.168.1.254;user= phone",response="61db6f5bf1b4ac464547bc3903ead800" ,nonce="02702734",algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTE R,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 354
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 12876 0 IN IP4 192.168.1.158
s=SIP Call
t=0 0
m=audio 23956 RTP/AVP 0 8 18 102 116 101
c=IN IP4 192.168.1.158
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (19 headers 16 lines) ---
Sending to 192.168.1.158:5061 (no NAT)
Using INVITE request as basis request - a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Found peer 'client-3' for 'client-3' from 192.168.1.158:50296
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 102
Found RTP audio format 116
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format L16 for ID 102
Found audio description format iLBC for ID 116
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x850c (ulaw|alaw|g729|ilbc|slin16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

[sunsetlive]

suite des logs

Peer audio RTP is at port 192.168.1.158:23956
Looking for 5 in users (domain 192.168.1.254)

<--- Reliably Transmitting (no NAT) to 192.168.1.158:5061 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bKfc3dd1e1;received =192.168.1.158
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>;tag=as7134625c
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.13.1~dfsg-1ubuntu2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Jul 3 08:52:31] NOTICE[14513]: chan_sip.c:22718 handle_request_invite: Call from 'client-3' (192.168.1.158:50296) to extension '5' rejected because extension not found in context 'users'.
Scheduling destruction of SIP dialog 'a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.158:52349 --->
ACK sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bKfc3dd1e1
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>;tag=as7134625c
Call-ID: a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158
Max-Forwards: 70
Date: Wed, 03 Jul 2013 06:52:20 GMT
CSeq: 102 ACK
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'OTlkMGE1ODgyM2FjNzYyNWIzMTFkN2QzOGE3ZjIyZTU.' Method: REGISTER
Really destroying SIP dialog '2db2795f36d6bb4733362aad3d972fe6@192.168.1.254:50 60' Method: BYE

<--- SIP read from UDP:192.168.1.150:5060 --->


<------------->
Really destroying SIP dialog 'a40cc394-a5620014-9e33763c-07952f4f@192.168.1.158' Method: ACK

<--- SIP read from UDP:192.168.1.150:5060 --->


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

<--- SIP read from UDP:192.168.1.150:5060 --->


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

<--- SIP read from UDP:192.168.1.150:5060 --->


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

<--- SIP read from UDP:192.168.1.150:5060 --->


<------------->
lol*CLI>

[jean]

on voit :

INVITE sip:client-1@192.168.1.150:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.254:5060;branch=z9hG4bK0bfe3f18;rport
Max-Forwards: 70
From: <sip:557@192.168.1.254:5060>;tag=as16824010
To: "client-1"<sip:client-1@192.168.1.254:5060>;tag=bb875e41

et

INVITE sip:5@192.168.1.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.158:5061;branch=z9hG4bK386bb6e8
From: "557" <sip:client-3@192.168.1.254>;tag=a40cc394a562001d5059681a-736ddf25
To: <sip:5@192.168.1.254>

le permier est un appel du poste 557 vers "client-1", et le second (répété dans les logs) est un appel de 557 vers "5"

Asterisk se comporte donc logiquement.

Quel client utilise tu ? Vérifie la config....