Hi
I use asterisk 13 and get sometimes this sip packet from my sip proxy server on 192.168.11.10 Asterisk found wrong device 1717 instance of proxy-10 :
<— SIP read from UDP:192.168.11.10:5060 —>
INVITE sip:36611950@192.168.11.10 SIP/2.0
Record-Route: sip:192.168.11.10;lr=on;ftag=as6a189055;nat=yes
Via: SIP/2.0/UDP 192.168.11.10;branch=z9hG4bK4c64.fa45eb0bdc6b846e8f4c97bd9a08aadc.0
Via: SIP/2.0/UDP 192.168.11.19:5060;received=192.168.11.19;branch=z9hG4bK6d0e0088;rport=5060
Max-Forwards: 69
From: “8144442860” sip:8144442860@192.168.11.19;tag=as6a189055
To: sip:36611950@192.168.11.10
Contact: sip:8144442860@192.168.11.19:5060
Call-ID: 3bec942504b2d72c645b177c3a71ddb7@192.168.11.19:5060
CSeq: 102 INVITE
User-Agent: FPBX-15.0.17.34(16.17.0)
Date: Sat, 26 Jun 2021 11:51:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 356
v=0
o=root 1801188738 1801188738 IN IP4 192.168.11.19
s=Asterisk PBX 16.17.0
c=IN IP4 192.168.11.19
t=0 0
m=audio 18724 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------->
[2021-06-26 16:23:09] VERBOSE[28735] chan_sip.c: — (16 headers 16 lines) —
[2021-06-26 16:23:09] VERBOSE[28735] chan_sip.c: Sending to 192.168.11.10:5060 (NAT)
[2021-06-26 16:23:09] VERBOSE[28735][C-00001466] chan_sip.c: Sending to 192.168.11.10:5060 (NAT)
[2021-06-26 16:23:09] VERBOSE[28735][C-00001466] chan_sip.c: Using INVITE request as basis request - 3bec942504b2d72c645b177c3a71ddb7@192.168.11.19:5060
[2021-06-26 16:23:09] VERBOSE[28735][C-00001466] chan_sip.c: Found peer ‘1717’ for ‘8144442860’ from 192.168.11.10:5060
[2021-06-26 16:23:09] VERBOSE[28735][C-00001466] netsock2.c: Using SIP RTP TOS bits 184
[2021-06-26 16:23:09] VERBOSE[28735][C-00001466] netsock2.c: Using SIP RTP CoS mark 5
[2021-06-26 16:23:09] VERBOSE[28735][C-00001466] chan_sip.c: Got SDP version 1801188738 and unique parts [root 1801188738 IN IP4 192.168.11.19]
[2021-06-26 16:23:09] VERBOSE[28735][C-00001466] chan_sip.c: Found RTP audio format 0
[2021-06-26 16:23:09] VERBOSE[28735][C-00001466] chan_sip.c: Found RTP audio format 8
[2021-06-26 16:23:09] VERBOSE[28735][C-00001466] chan_sip.c: Found RTP audio format 3
here is currect device matching :
<— SIP read from UDP:192.168.11.10:5060 —>
INVITE sip:36611950@192.168.11.10 SIP/2.0
Record-Route: sip:192.168.11.10;lr=on;ftag=as1fe60eae;nat=yes
Via: SIP/2.0/UDP 192.168.11.10;branch=z9hG4bKb879.9b1f273233d8f3c61502696a8a88930f.0
Via: SIP/2.0/UDP 192.168.11.19:5060;received=192.168.11.19;branch=z9hG4bK7fcc97ae;rport=5060
Max-Forwards: 69
From: “32576548” sip:32576548@192.168.11.19;tag=as1fe60eae
To: sip:36611950@192.168.11.10
Contact: sip:32576548@192.168.11.19:5060
Call-ID: 08ed4f5778245cfb7996de0916a4ace7@192.168.11.19:5060
CSeq: 102 INVITE
User-Agent: FPBX-15.0.17.34(16.17.0)
Date: Sat, 26 Jun 2021 11:47:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 354
v=0
o=root 505936696 505936696 IN IP4 192.168.11.19
s=Asterisk PBX 16.17.0
c=IN IP4 192.168.11.19
t=0 0
m=audio 15398 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------->
[2021-06-26 16:18:42] VERBOSE[28735] chan_sip.c: — (16 headers 16 lines) —
[2021-06-26 16:18:42] VERBOSE[28735] chan_sip.c: Sending to 192.168.11.10:5060 (NAT)
[2021-06-26 16:18:42] VERBOSE[28735][C-00001465] chan_sip.c: Sending to 192.168.11.10:5060 (NAT)
[2021-06-26 16:18:42] VERBOSE[28735][C-00001465] chan_sip.c: Using INVITE request as basis request - 08ed4f5778245cfb7996de0916a4ace7@192.168.11.19:5060
[2021-06-26 16:18:42] VERBOSE[28735][C-00001465] chan_sip.c: Found peer ‘proxy-10’ for ‘32576548’ from 192.168.11.10:5060
[2021-06-26 16:18:42] VERBOSE[28735][C-00001465] netsock2.c: Using SIP RTP TOS bits 184
[2021-06-26 16:18:42] VERBOSE[28735][C-00001465] netsock2.c: Using SIP RTP CoS mark 5
[2021-06-26 16:18:42] VERBOSE[28735][C-00001465] chan_sip.c: Got SDP version 505936696 and unique parts [root 505936696 IN IP4 192.168.11.19]
[2021-06-26 16:18:42] VERBOSE[28735][C-00001465] chan_sip.c: Found RTP audio format 0
[2021-06-26 16:18:42] VERBOSE[28735][C-00001465] chan_sip.c: Found RTP audio format 8
here is 1717 and proxy-10 sip settings:
[proxy-10]
host=192.168.11.10
type=peer
context=from-trunk
qualify=yes
[1717]
deny=0.0.0.0/0.0.0.0
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
defaultuser=
trustrpid=yes
user_eq_phone=no
sendrpid=pai
type=friend
session-timers=accept
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/1717
accountcode=
permit=192.168.11.20/255.255.255.128
recordonfeature=apprecord
recordofffeature=apprecord
callcounter=yes
faxdetect=no
how does that happen?