Bug #2671
closedOsmoBTS + asterisk
0%
Description
good day colleagues,
I have found a bug, when tested openbts + asterisk
when OBTS send invite to asterisk (I use asterisk on outside server), i have localhost ip in section body option message invite
its dump
@
<--- SIP read from UDP:10.135.12.4:5069 --->
INVITE sip:111@10.135.12.38:5060 SIP/2.0
Via: SIP/2.0/UDP 10.135.12.4:5069;rport;branch=z9hG4bKDF1N7Zt4vecjN
Max-Forwards: 70
From: <sip:102@10.135.12.4:5069>;tag=8DFF394y7pmjH
To: <sip:111@10.135.12.38:5060>
Call-ID: 5c402941-493c-1236-e1af-3860773e9612
CSeq: 930060148 INVITE
Contact: <sip:10.135.12.4:5069>
User-Agent: sofia-sip/1.12.11devel
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Length: 124
v=0
o=Osmocom 0 0 IN IP4 127.0.0.1
s=GSM Call
c=IN IP4 127.0.0.1
t=0 0
m=audio 40998 RTP/AVP 98
a=rtpmap:98 AMR/8000
<------------->
--- (13 headers 7 lines) ---
Sending to 10.135.12.4:5069 (no NAT)
Sending to 10.135.12.4:5069 (no NAT)
Using INVITE request as basis request - 5c402941-493c-1236-e1af-3860773e9612
Found peer 'GSM' for '102' from 10.135.12.4:5069
== Using SIP RTP CoS mark 5
Found RTP audio format 98
Found audio description format AMR for ID 98
Capabilities: us - (gsm|g722|amr), peer - audio=(amr)/video=(nothing)/text=(nothing), combined - (amr)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 127.0.0.1:40998
Looking for 111 in gsmsubscriber (domain 10.135.12.38)
sip_route_dump: route/path hop: <sip:10.135.12.4:5069>
@
and sent RTP packet to 127.0.0.1:40998
