Hi guys,
I’m setting up (for a 2nd time) my asteriskNow server. I have a few SIP phones with a single trunk to my SIP Provider. The first time around, a few months back, it all worked out fine.
This time, my inbound calls are fine, but I can’t generate outbound calls.
The problem I see is that my outbound calls INVITE Request-URI Host has my own IP address (96.71.145.239) instead of my Provider’s (159.18.161.67):
INVITE sip:6478691125@96.71.145.239:5060 SIP/2.0
So when my INVITE reaches the Provider, they route the invite right back to me, instead of connecting my call.
No. Time Source Destination Protocol Info
4 *REF* 96.71.145.239 159.18.161.67 SIP/SDP Request: INVITE sip:6478691125@96.71.145.239:5060, with session description
5 0.065184 159.18.161.67 96.71.145.239 SIP Status: 100 Giving a try
6 0.069175 159.18.161.67 96.71.145.239 SIP/SDP Request: INVITE sip:6478691125@96.71.145.239:5060, with session description
7 0.069356 96.71.145.239 159.18.161.67 SIP Status: 100 Trying
8 0.070812 96.71.145.239 159.18.161.67 SIP Request: CANCEL sip:6478691125@96.71.145.239:5060
9 0.141846 159.18.161.67 96.71.145.239 SIP Request: CANCEL sip:6478691125@96.71.145.239:5060
10 0.141987 96.71.145.239 159.18.161.67 SIP Status: 487 Request Terminated
11 0.142039 96.71.145.239 159.18.161.67 SIP Status: 200 OK
12 0.142573 159.18.161.67 96.71.145.239 SIP Status: 200 canceling
13 1.141913 96.71.145.239 159.18.161.67 SIP Status: 487 Request Terminated
The rest of the SIP packet looks just right. The message header To: section is correct and has my Provider’s IP:
To: <sip:6478691125@159.18.161.67>
SIP to address: sip:6478691125@159.18.161.67
SIP to address User Part: 6478691125
SIP to address Host Part: 159.18.161.67
Why would the Request-URI differ from the To: header? What I’m I configuring wrong or not configuring?
Its a pretty basic installation.
All I have for my trunk config is (which worked the first time):
disallow=all
allow=ulaw
dtmfmode=rfc2833
fromdomain=vp.thinktel.ca
host=159.18.161.67
insecure=invite,port
type=friend
nat=yes
The full text of the an initial INVITE decoded packet follows:
No. Time Source Destination Protocol Info
13 6.307536 96.71.145.239 159.18.161.67 SIP/SDP Request: INVITE sip:6478691125@96.71.145.239:5060, with session description
Frame 13 (828 bytes on wire, 828 bytes captured)
Arrival Time: Oct 30, 2010 14:53:42.786416000
[Time delta from previous captured frame: 1.869409000 seconds]
[Time delta from previous displayed frame: 2.873301000 seconds]
[Time since reference or first frame: 6.307536000 seconds]
Frame Number: 13
Frame Length: 828 bytes
Capture Length: 828 bytes
[Frame is marked: False]
[Protocols in frame: sll:ip:udp:sip:sdp]
[Coloring Rule Name: UDP]
[Coloring Rule String: udp]
Linux cooked capture
Packet type: Sent by us (4)
Link-layer address type: 512
Link-layer address length: 0
Source: <MISSING>
Protocol: IP (0x0800)
Internet Protocol, Src: 96.71.145.239 (96.71.145.239), Dst: 159.18.161.67 (159.18.161.67)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x60 (DSCP 0x18: Class Selector 3; ECN: 0x00)
0110 00.. = Differentiated Services Codepoint: Class Selector 3 (0x18)
.... ..0. = ECN-Capable Transport (ECT): 0
.... ...0 = ECN-CE: 0
Total Length: 812
Identification: 0x3ac0 (15040)
Flags: 0x00
0.. = Reserved bit: Not Set
.0. = Don't fragment: Not Set
..0 = More fragments: Not Set
Fragment offset: 0
Time to live: 64
Protocol: UDP (0x11)
Header checksum: 0x1bb1 [correct]
[Good: True]
[Bad : False]
Source: 96.71.145.239 (96.71.145.239)
Destination: 159.18.161.67 (159.18.161.67)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Source port: sip (5060)
Destination port: sip (5060)
Length: 792
Checksum: 0x35e9 [validation disabled]
[Good Checksum: False]
[Bad Checksum: False]
Session Initiation Protocol
Request-Line: INVITE sip:6478691125@96.71.145.239:5060 SIP/2.0
Method: INVITE
Request-URI: sip:6478691125@96.71.145.239:5060
Request-URI User Part: 6478691125
Request-URI Host Part: 96.71.145.239
Request-URI Host Port: 5060
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 96.71.145.239:5060;branch=z9hG4bK11cac908;rport
Transport: UDP
Sent-by Address: 96.71.145.239
Sent-by port: 5060
Branch: z9hG4bK11cac908
RPort: rport
From: "6473172818" <sip:6473172818@vp.thinktel.ca>;tag=as76a0018a
SIP Display info: "6473172818"
SIP from address: sip:6473172818@vp.thinktel.ca
SIP from address User Part: 6473172818
SIP from address Host Part: vp.thinktel.ca
SIP tag: as76a0018a
To: <sip:6478691125@159.18.161.67>
SIP to address: sip:6478691125@159.18.161.67
SIP to address User Part: 6478691125
SIP to address Host Part: 159.18.161.67
Contact: <sip:6473172818@96.71.145.239>
Contact Binding: <sip:6473172818@96.71.145.239>
URI: <sip:6473172818@96.71.145.239>
SIP contact address: sip:6473172818@96.71.145.239
Call-ID: 0230a1f07327c4323b4e96dc630a6f3f@vp.thinktel.ca
CSeq: 102 INVITE
Sequence Number: 102
Method: INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 30 Oct 2010 18:53:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 215
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 5212 5212 IN IP4 96.71.145.239
Owner Username: root
Session ID: 5212
Session Version: 5212
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 96.71.145.239
Session Name (s): session
Connection Information (c): IN IP4 96.71.145.239
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 96.71.145.239
Time Description, active time (t): 0 0
Session Start Time: 0
Session Stop Time: 0
Media Description, name and address (m): audio 16720 RTP/AVP 0 101
Media Type: audio
Media Port: 16720
Media Protocol: RTP/AVP
Media Format: ITU-T G.711 PCMU
Media Format: DynamicRTP-Type-101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute Fieldname: rtpmap
Media Format: 0
MIME Type: PCMU
Sample Rate: 8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Format: 101
MIME Type: telephone-event
Sample Rate: 8000
Media Attribute (a): fmtp:101 0-16
Media Attribute Fieldname: fmtp
Media Format: 101 [telephone-event]
Media format specific parameters: 0-16
Media Attribute (a): ptime:20
Media Attribute Fieldname: ptime
Media Attribute Value: 20
Media Attribute (a): sendrecvnote: My own IP and telephone numbers have been changed for privacy reasons.
edit: I’m using asteriskNow 1.7.1-2: Asterisk 1.4.36-1 and freepbx 2.7.0-7 CentOS 5.5
Your help is appreciated.
Thanks
Felix