Hi,
I have an Asterisk server NATed and some SIP Speakers are connecting to the PBX.
When I initiate a call, the RTP packets are sending over to the local IP address of the SIP Speaker.
Here’s the malformed packet:
Frame 4: 886 bytes on wire (7088 bits), 886 bytes captured (7088 bits)
Ethernet II, Src: Fortinet_84:aa:d0 (08:5b:0e:84:aa:d0), Dst: VMware_7e:11:4d (00:0c:29:7e:11:4d)
Internet Protocol Version 4, Src: 204.101.58.123, Dst: 172.29.9.94
User Datagram Protocol, Src Port: 3212, Dst Port: 5060
Session Initiation Protocol (200)
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP 66.158.139.50:5060;rport=5060;received=66.158.139.50;branch=z9hG4bK0675f91f
Call-ID: 218acd6227a6a2c55cd2fd056e63b606@66.158.139.50:5060
[Generated Call-ID: 218acd6227a6a2c55cd2fd056e63b606@66.158.139.50:5060]
From: “Simon-Phone” sip:207@66.158.139.50 ;tag=as25910f6b
To: sip:370@204.101.58.123;ob ;tag=so2xX6QZL9M30CuZSENAjMR5UPkhrEGT
CSeq: 102 INVITE
Contact: sip:370@204.101.58.123:3212;ob
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 279
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 3790112006 3790112007 IN IP4 192.168.44.64
Session Name (s): pjmedia
Bandwidth Information (b): AS:699
Time Description, active time (t): 0 0
Session Attribute (a): X-nat:0
Media Description, name and address (m): audio 4026 RTP/AVP 0 101
Connection Information ©: IN IP4 192.168.44.64
Bandwidth Information (b): TIAS:650000
Media Attribute (a): rtcp:4027 IN IP4 192.168.44.64
Media Attribute (a): sendrecv
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-16
[Generated Call-ID: 218acd6227a6a2c55cd2fd056e63b606@66.158.139.50:5060]
The address in the message body is wrong but “To:” in the header is the right address.
We have like 30 SIP Speakers connected to the PBX and it was working fine before.
Thanks for any response.
chan_sip or pjsip channel driver ?
I tried both but still the same issue.
Show sip settings for endpoint configuration
Here is the extension configuration
[370]
deny=0.0.0.0/0.0.0.0
disallow=all
secret=6755f8f28a90d33bebee031ca46162c7
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
defaultuser=
trustrpid=no
sendrpid=no
type=peer
session-timers=accept
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
encryption=no
namedcallgroup=
namedpickupgroup=
allow=alaw;ulaw
dial=SIP/370
accountcode=
permit=0.0.0.0/0.0.0.0
callerid=Ford Lincoln 440 <370>
callcounter=yes
faxdetect=no
And the SIP show peer
Name : 370
Description :
Secret :
MD5Secret :
Remote Secret:
Context : from-internal
Record On feature : automon
Record Off feature : automon
Subscr.Cont. :
Language : en
Tonezone :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox :
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : “Ford Lincoln 440” <370>
MaxCallBR : 384 kbps
Expire : 105
Insecure : no
Force rport : Yes
Symmetric RTP: Yes
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 204.101.58.123:3235
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 370
SIP Options : (none)
Codecs : (alaw)
Codec Order : (alaw:20)
Auto-Framing : No
Status : OK (8 ms)
Useragent : AXIS C3003-E Network Horn Speaker
Reg. Contact : sip:370@204.101.58.123:3235;ob
Qualify Freq : 60000 ms
Keepalive : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
add to the general section the following parameters localnet , media_addr and extern_addr
That have been done but no chance.
groupedci:
192.168.44.x
If this is part of the local net add it also to the SIP configuraton
This is the local address of the SIP Speaker.
This one is supposed to be connecting to our public IP addess but the connection information is not translated and traffic is being sent to the local IP address instead of the public IP address as it supposed to.
Check router setting and device SIP setting, mybe you re missing a NAT configuration on the device
system
closed
March 11, 2020, 2:04am
#11
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