[Jun 8 11:46:32] VERBOSE[17509] logger.c:
<--- SIP read from IP_US_Provider:5070 --->
INVITE sip:99990900448708757743@IP_mein_Asterisk SIP/2.0
To: 99990900448708757743<sip:99990900448708757743@IP_mein_Asterisk>
From: 1000<sip:1000@IP_mein_Asterisk>;tag=2e02b895
Via: SIP/2.0/UDP IP_US_Provider:5070;branch=z9hG4bK-c9a96c173a65a1c6c7b55e830117c83c;rport
Call-ID: c9a96c173a65a1c6c7b55e830117c83c
CSeq: 1 INVITE
Contact: <sip:1000@IP_US_Provider:5070>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE
User-Agent: sipcli/v1.8
Content-Type: application/sdp
Content-Length: 281
v=0
o=sipcli-Session 2061128232 1612946516 IN IP4 IP_US_Provider
s=sipcli
c=IN IP4 IP_US_Provider
t=0 0
m=audio 5075 RTP/AVP 18 0 8 101
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
[Jun 8 11:46:32] VERBOSE[17509] logger.c: --- (12 headers 13 lines) ---
[Jun 8 11:46:32] VERBOSE[17509] logger.c: Sending to IP_US_Provider : 5070 (no NAT)
[Jun 8 11:46:32] VERBOSE[17509] logger.c: Using INVITE request as basis request - c9a96c173a65a1c6c7b55e830117c83c
[Jun 8 11:46:32] VERBOSE[17509] logger.c: Found no matching peer or user for 'IP_US_Provider:5070'
[Jun 8 11:46:32] VERBOSE[17509] logger.c: Found RTP audio format 18
[Jun 8 11:46:32] VERBOSE[17509] logger.c: Found RTP audio format 0
[Jun 8 11:46:32] VERBOSE[17509] logger.c: Found RTP audio format 8
[Jun 8 11:46:32] VERBOSE[17509] logger.c: Found RTP audio format 101
[Jun 8 11:46:32] VERBOSE[17509] logger.c: Found audio description format G729 for ID 18
[Jun 8 11:46:32] VERBOSE[17509] logger.c: Found audio description format PCMU for ID 0
[Jun 8 11:46:32] VERBOSE[17509] logger.c: Found audio description format PCMA for ID 8
[Jun 8 11:46:32] VERBOSE[17509] logger.c: Found audio description format telephone-event for ID 101
[Jun 8 11:46:32] VERBOSE[17509] logger.c: Capabilities: us - 0xe0e (gsm|ulaw|alaw|g726|speex|ilbc), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Jun 8 11:46:32] VERBOSE[17509] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Jun 8 11:46:32] DEBUG[17509] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0)
[Jun 8 11:46:32] VERBOSE[17509] logger.c: Peer audio RTP is at port IP_US_Provider:5075
[Jun 8 11:46:32] VERBOSE[17509] logger.c: Looking for 99990900448708757743 in default (domain IP_mein_Asterisk)
[Jun 8 11:46:32] VERBOSE[17509] logger.c:
<--- Reliably Transmitting (NAT) to IP_US_Provider:5070 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP IP_US_Provider:5070;branch=z9hG4bK-c9a96c173a65a1c6c7b55e830117c83c;received=IP_US_Provider;rport=5070
From: 1000<sip:1000@IP_mein_Asterisk>;tag=2e02b895
To: 99990900448708757743<sip:99990900448708757743@IP_mein_Asterisk>;tag=as736933bd
Call-ID: c9a96c173a65a1c6c7b55e830117c83c
CSeq: 1 INVITE
User-Agent: MyDevice
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0