On 01/22/11 20:00, Da Rock wrote:
On 01/21/11 20:28, Da Rock wrote:
On 01/21/11 03:19, Tom Rymes wrote:
On 01/19/2011 10:34 PM, Da Rock wrote:
WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to
non-existing call leg on other UA. SIP dialog
'481cf0543743e6bb7006991d409ed3bc@150.101.178.33:5060'. Giving up.
Have you tried disallowing re-invites?
Sorry for the delay, but I've tried both yes and no- one of the
first things I tried, but I get your reasoning.
Thanks
Some more information has come to light- bit of luck this clue
happen to come to my attention: My provider could be using a
Broadworks system. Does that change things much?
In my sip debug for the peer it flashed up realm="Broadworks" from
the peer.
Being very new to asterisk and SIP I'm still trying to learn the
"protocol". Perhaps someone here may be able to correct my
understanding if necessary (and a point in the right direction would
help significantly).
What I wasn't realising was that if I set sip debug on it output the
entire sip message. So my output looks like this:
-- Executing [0871271201@users:1] Goto("SIP/<local ata>-00000017",
"internode-outgoing,0871271201,1") in new stack
-- Goto (internode-outgoing,0871271201,1)
-- Executing [0871271201@internode-outgoing:1] Dial("SIP/<local
ata>-00000017", "SIP/0871271201@sip-out") in new stack
Audio is at 5060
Video is at <asterisk ip>:5060
Text is at <asterisk ip>:5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding video codec 0x100000 (h263p) to SDP
Adding text codec 0x4000000 (red) to SDP
Adding text codec 0x8000000 (t140) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 203.2.134.1:5060:
INVITE sip:0871271...@sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK41104eea
Max-Forwards: 70
From: "Skinner's Home" <sip:<local ata>@<asterisk ip>>;tag=as6683ffea
To: <sip:0871271...@sip.internode.on.net>
Contact: <sip:<local ata>@<asterisk ip>:5060>
Call-ID: 5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.1.1
Date: Sat, 22 Jan 2011 11:23:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 736
v=0
o=root 189870721 189870721 IN IP4 <asterisk ip>
s=Asterisk PBX 1.8.1.1
c=IN IP4 <asterisk ip>
b=CT:384
t=0 0
m=audio 17220 RTP/AVP 0 3 8 112 5 10 7 110 97 111 9 118 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a= -- Called 0871271201@sip-out
<--- SIP read from UDP:203.2.134.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <asterisk ip>:5060;received=<my ext
ip>;branch=z9hG4bK41104eea;rport=61533
From: "<local ata>" <sip:<local ata>@<asterisk ip>>;tag=as6683ffea
To: <sip:0871271...@sip.internode.on.net>
Call-ID: 5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
Reliably Transmitting (no NAT) to 203.2.134.1:5060:
OPTIONS sip:sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK7ab229d7
Max-Forwards: 70
From: "Unknown" <sip:Unknown@<asterisk ip>>;tag=as72e93b63
To: <sip:sip.internode.on.net>
Contact: <sip:Unknown@<asterisk ip>:5060>
Call-ID: 068700ad12b1d48a5059a1837a04a8b3@<asterisk ip>:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.1.1
Date: Sat, 22 Jan 2011 11:24:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:203.2.134.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <asterisk ip>:5060;received=<my ext
ip>;branch=z9hG4bK7ab229d7;rport=61533
From: "Unknown" <sip:Unknown@<asterisk ip>>;tag=as72e93b63
To: <sip:sip.internode.on.net>;tag=488684762-1295695493625
Call-ID: 068700ad12b1d48a5059a1837a04a8b3@<asterisk ip>:5060
CSeq: 102 OPTIONS
Allow-Events:
call-info,line-seize,dialog,message-summary,as-feature-event
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog
'068700ad12b1d48a5059a1837a04a8b3@<asterisk ip>:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 203.2.134.1:5060:
OPTIONS sip:sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK53bdf76d
Max-Forwards: 70
From: "Unknown" <sip:Unknown@<asterisk ip>>;tag=as2a328631
To: <sip:sip.internode.on.net>
Contact: <sip:Unknown@<asterisk ip>:5060>
Call-ID: 42d9681e4b8dcad66879b54c697d7e32@<asterisk ip>:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.1.1
Date: Sat, 22 Jan 2011 11:24:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:203.2.134.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <asterisk ip>:5060;received=<my ext
ip>;branch=z9hG4bK53bdf76d;rport=61533
From: "Unknown" <sip:Unknown@<asterisk ip>>;tag=as2a328631
To: <sip:sip.internode.on.net>;tag=1907972657-1295695493739
Call-ID: 42d9681e4b8dcad66879b54c697d7e32@<asterisk ip>:5060
CSeq: 102 OPTIONS
Allow-Events:
call-info,line-seize,dialog,message-summary,as-feature-event
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog
'42d9681e4b8dcad66879b54c697d7e32@<asterisk ip>:5060' Method: OPTIONS
<--- SIP read from UDP:203.2.134.1:5060 --->
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP <asterisk ip>:5060;received=<my ext
ip>;branch=z9hG4bK41104eea;rport=61533
From: "<local ata>" <sip:<local ata>@<asterisk ip>>;tag=as6683ffea
To: <sip:0871271...@sip.internode.on.net>;tag=aprqngfrt-3s5u6r20000c6
Call-ID: 5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
[Jan 22 21:24:08] WARNING[993]: chan_sip.c:19069
handle_response_invite: Re-invite to non-existing call leg on other
UA. SIP dialog '5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060'.
Giving up.
Transmitting (no NAT) to 203.2.134.1:5060:
ACK sip:0871271...@sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK41104eea
Max-Forwards: 70
From: "<local ata>" <sip:<local ata>@<asterisk ip>>;tag=as6683ffea
To: <sip:0871271...@sip.internode.on.net>;tag=aprqngfrt-3s5u6r20000c6
Contact: <sip:<local ata>@<asterisk ip>:5060>
Call-ID: 5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.1.1
Content-Length: 0
---
Scheduling destruction of SIP dialog
'5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060' in 6400 ms
(Method: INVITE)
-- SIP/sip-out-00000018 is circuit-busy
Scheduling destruction of SIP dialog
'5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060' in 6400 ms
(Method: INVITE)
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [0871271201@internode-outgoing:2] Answer("SIP/<local
ata>-00000017", "2") in new stack
-- Executing [0871271201@internode-outgoing:3]
Playback("SIP/<local ata>-00000017", "ss-noservice") in new stack
-- <SIP/<local ata>-00000017> Playing 'ss-noservice.gsm'
(language 'en')
== Spawn extension (internode-outgoing, 0871271201, 3) exited
non-zero on 'SIP/<local ata>-00000017'
Really destroying SIP dialog
'5cf55909146ff03907fcc86437809bcc@<asterisk ip>:5060' Method: INVITE
[Jan 22 21:24:36] NOTICE[993]: chan_sip.c:12142 sip_reregister: --
Re-registration for 0731292...@sip.internode.on.net
> doing dnsmgr_lookup for 'sip.internode.on.net'
> ast_get_srv: SRV lookup for '_sip._udp.sip.internode.on.net' mapped
to host sip.internode.on.net, port 5060
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 203.2.134.1:5060:
REGISTER sip:sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK07b4e860
Max-Forwards: 70
From: <sip:0731292...@sip.internode.on.net>;tag=as1f1ee1a8
To: <sip:0731292...@sip.internode.on.net>
Call-ID: 597b79434d3c224a3b484fd718502b34@<asterisk ip>
CSeq: 1440 REGISTER
User-Agent: Asterisk PBX 1.8.1.1
Authorization: Digest username="0731292848", realm="BroadWorks",
algorithm=MD5, uri="sip:sip.internode.on.net",
nonce="BroadWorksXgj5g2j3vTbun57kBW",
response="4a529cc44fa4c19924fed72fed1da2e2", qop=auth,
cnonce="44b80039", nc=0000053a
Expires: 120
Contact: <sip:0731292848@<asterisk ip>:5060>
Content-Length: 0
---
<--- SIP read from UDP:203.2.134.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <asterisk ip>:5060;received=<my ext
ip>;branch=z9hG4bK07b4e860;rport=61533
From: <sip:0731292...@sip.internode.on.net>;tag=as1f1ee1a8
To: <sip:0731292...@sip.internode.on.net>;tag=aprqcauh8h3-7m1v3j200810a
Call-ID: 597b79434d3c224a3b484fd718502b34@<asterisk ip>
CSeq: 1440 REGISTER
Contact: <sip:0731292848@<asterisk ip>:5060>;expires=150
<------------->
--- (7 headers 0 lines) ---
Scheduling destruction of SIP dialog
'597b79434d3c224a3b484fd718502b34@<asterisk ip>' in 6400 ms (Method:
REGISTER)
[Jan 22 21:24:36] NOTICE[993]: chan_sip.c:19502
handle_response_register: Outbound Registration: Expiry for
sip.internode.on.net is 150 sec (Scheduling reregistration in 135 s)
From this it looks to me that the call is gong through, but asterisk
is not acting on it- is that right? Or am I misreading it?
I see the outgoing call and an sip message generated and sent the
provider, the provider calls back trying, and then sends 2x 200
messages saying ok. The provider then gets no response and sends back
a 408. And then asterisk acts on it and tries to attach to a non
existent leg.
Internode insist on simply opening up the firewall on a dmz to allow
complete access to the asterisk server with theirs, but I have 2
points to contend with that:
1. What about spoofing? They may have HA cluster attached to that IP,
but what about an attack on my server? Couldn't someone spoof the IP
for their own purposes? (Or I could be paranoid too :) )
2. I aim to setup a hosting solution, so I can allow clients to peer
with my asterisk (not using my trunk though) so I need to be security
conscious, and allow more than just the nodephone server to connect.
Any help?
Cheers
