Trunk con SBC Fastweb

[Tipdrill]

Buongiorno,
ho un problema con un sbc fastweb in collaudo.
Mi hanno assegnato un numero provvisorio per i test, in uscita nessun problema in ingresso ricevo numero inesistente. Lato loro dicono sia tutto ok.
Configurando il DID sul mio gateway ricevo numero inesistente, se non configuro nessun DID la chiamata cade senza nessun avviso.

Ricevo questo errore ad ogni tentativo di chiamata in ingresso:
 

<--- SIP read from UDP:10.10.1.4:5060 --->
INVITE sip:+3907*****@172.16.51.10:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.1.4:5060;branch=z9hG4bK3f8ltj10b8dpajcf6tf0.1
Call-ID: 2c9fe8ae09f6da6c4c6db467dbcba1a2@10.247.30.167
CSeq: 1 INVITE
To: <sip:+3907*****@172.16.51.10;lr;user=phone>
Content-Type: application/sdp
Max-Forwards: 58
Supported: 100rel
Allow: UPDATE,NOTIFY,ACK,SUBSCRIBE,INVITE,INFO,REFER,MESSAGE,CANCEL,PRACK,BYE,OPTIONS
Contact: <sip:392*****@10.10.1.4:5060;transport=udp>
From: <sip:392*****@10.10.1.4;user=phone>;tag=7150b724
Content-Length: 270

v=0
o=HPE-AS 58609 1 IN IP4 10.10.1.12
s=IMSS
c=IN IP4 10.10.1.12
t=0 0
a=sendrecv
m=audio 10014 RTP/AVP 18 101
b=RR:0
b=RS:0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:40
a=sqn:0
a=cdsc:1 image udptl t38
a=ptime:20

<------------->
--- (12 headers 16 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 4
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Sending to 10.10.1.4 : 5060 (NAT)
Using INVITE request as basis request - 2c9fe8ae09f6da6c4c6db467dbcba1a2@10.247.30.167
[2024-01-09 10:19:57] NOTICE[1537]: chan_sip.c:15025 check_peer_ok: [YEASTARDNSMAP] Get peer by ip and toexten:+3907*****
Found peer 'trunk-sps-Unigate' for '392*****' from 10.10.1.4:5060
Found RTP audio format 18
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0x90e (gsm|ulaw|alaw|g726|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.10.1.12:10014

Looking for +3907***** in DID_inbound_trunk-sps-Unigate (domain 172.16.51.10)

 

<--- Reliably Transmitting (NAT) to 10.10.1.4:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.10.1.4:5060;branch=z9hG4bK3f8ltj10b8dpajcf6tf0.1;received=10.10.1.4
From: <sip:392*****@10.10.1.4;user=phone>;tag=7150b724
To: <sip:+3907*****@172.16.51.10;lr;user=phone>;tag=as6499b9a0
Call-ID: 2c9fe8ae09f6da6c4c6db467dbcba1a2@10.247.30.167
CSeq: 1 INVITE
Server: TE100
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

 

[2024-01-09 10:19:57] NOTICE[1537]: chan_sip.c:21279 handle_request_invite: Call from 'trunk-sps-Unigate' to extension '+3907*****' rejected because extension not found.

Scheduling destruction of SIP dialog '2c9fe8ae09f6da6c4c6db467dbcba1a2@10.247.30.167' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:10.10.1.4:5060 --->
ACK sip:+3907*****@172.16.51.10:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.10.1.4:5060;branch=z9hG4bK3f8ltj10b8dpajcf6tf0.1
CSeq: 1 ACK
Call-ID: 2c9fe8ae09f6da6c4c6db467dbcba1a2@10.247.30.167
To: <sip:+3907*****9@172.16.51.10;lr;user=phone>;tag=as6499b9a0
Max-Forwards: 58
From: <sip:392*****@10.10.1.4;user=phone>;tag=7150b724
Content-Length: 0

Grazie

[Andrea Annoni]

Sicuro di avere i codec corretti?

[Tipdrill]

Ciao Andrea,

si, a cascata ulaw - alaw - g729 - gsm, di base l'sbc utilizza g729, ho fatto vari test e senza quest'ultimo non va nemmeno l'outbound.