Das ging ja schnell.
Zur info: 192.168.2.11 ist die Mobydick, 192.168.2.246 die OXO.
Danke
<-- SIP read from 192.168.2.246:1042:
INVITE sip:80@192.168.2.11;user=phone SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE
Supported: 100rel,from-change,timer
User-Agent: OxO_GW_710/115.006
Session-Expires: 43200
P-Asserted-Identity: “Vorname Nachname” <sip:49000888815@192.168.2.246;user=phone>
To: <sip:80@192.168.2.11;user=phone>
From: “Vorname Nachname” <sip:49000888815@192.168.2.246;user=phone>;tag=4d15cfd48dab3bffb6385a3faad481e0
Contact: “Vorname Nachname” <sip:49000888815@192.168.2.246;transport=UDP;user=phone>
Content-Type: application/sdp
Call-ID: 22fd601bdc3a5844a2bb144dd4387912@192.168.2.246
CSeq: 14214636 INVITE
Via: SIP/2.0/UDP 192.168.2.246;rport;branch=z9hG4bKe94905c5ec4da57643da71de996ab75c
Max-Forwards: 70
Content-Length: 272
v=0
o=default 1292083681 1292083681 IN IP4 0.0.0.0
s=-
c=IN IP4 192.168.2.246
t=0 0
m=audio 32000 RTP/AVP 18 106 4 8 0
a=sendrecv
a=fmtp:18 annexb=no
a=rtpmap:106 telephone-event/8000
a=fmtp:106 0-15
a=fmtp:4 annexa=no
a=maxptime:90
a=silenceSupp:off - - - -
— (15 headers 13 lines) —
Using INVITE request as basis request - 22fd601bdc3a5844a2bb144dd4387912@192.168.2.246
Sending to 192.168.2.246 : 5060 (NAT)
Found no matching peer or user for ‘192.168.2.246:1042’
Found RTP audio format 18
Found RTP audio format 106
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 0
Peer audio RTP is at port 192.168.2.246:32000
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 80 in no-auth-in (domain 192.168.2.11;user=phone)
list_route: hop: <sip:49000888815@192.168.2.246;transport=UDP;user=phone>
Transmitting (NAT) to 192.168.2.246:1042:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.246;branch=z9hG4bKe94905c5ec4da57643da71de996ab75c;received=192.168.2.246;rport=1042
From: “Vorname Nachname” <sip:49000888815@192.168.2.246;user=phone>;tag=4d15cfd48dab3bffb6385a3faad481e0
To: <sip:80@192.168.2.11;user=phone>
Call-ID: 22fd601bdc3a5844a2bb144dd4387912@192.168.2.246
CSeq: 14214636 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:80@192.168.2.11>
Content-Length: 0
-- Executing Macro("SIP/192.168.2.246-081d67c0", "mdc_emergency|80") in new stack
-- Executing NoOp("SIP/192.168.2.246-081d67c0", "macro-mdc_emergency:: Exten: 80") in new stack
-- Executing Goto("SIP/192.168.2.246-081d67c0", "80|1") in new stack
-- Goto (macro-mdc_emergency,80,1)
-- Executing NoOp("SIP/192.168.2.246-081d67c0", "80 ist keine Notrufnummer") in new stack
-- Executing Hangup("SIP/192.168.2.246-081d67c0", "0") in new stack
== Spawn extension (no-auth-in, 80, 2) exited non-zero on ‘SIP/192.168.2.246-081d67c0’
Scheduling destruction of call ‘22fd601bdc3a5844a2bb144dd4387912@192.168.2.246’ in 32000 ms
Reliably Transmitting (NAT) to 192.168.2.246:1042:
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP 192.168.2.246;branch=z9hG4bKe94905c5ec4da57643da71de996ab75c;received=192.168.2.246;rport=1042
From: “Vorname Nachname” <sip:49000888815@192.168.2.246;user=phone>;tag=4d15cfd48dab3bffb6385a3faad481e0
To: <sip:80@192.168.2.11;user=phone>;tag=as4a8c46e5
Call-ID: 22fd601bdc3a5844a2bb144dd4387912@192.168.2.246
CSeq: 14214636 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
mobydick*CLI>
<-- SIP read from 192.168.2.246:1042:
ACK sip:80@192.168.2.11;user=phone SIP/2.0
Call-ID: 22fd601bdc3a5844a2bb144dd4387912@192.168.2.246
From: “Vorname Nachname” <sip:49000888815@192.168.2.246;user=phone>;tag=4d15cfd48dab3bffb6385a3faad481e0
To: <sip:80@192.168.2.11;user=phone>;tag=as4a8c46e5
Via: SIP/2.0/UDP 192.168.2.246;rport;branch=z9hG4bKe94905c5ec4da57643da71de996ab75c
CSeq: 14214636 ACK
Content-Length: 0
— (7 headers 0 lines) —