Surely there must be someone here who can help me with this problem.
I have spent weeks trying to get this damned service to work with no luck.
I have incoming calls working, but no outgoing. If get outgoing working
then incoming don't work.
I have sent this problem to this list a couple of times with little or no
response, and I _really_ need some help to sort it out.
I have an asterisk 1.8 server running on FreeBSD 8.1, and another FreeBSD
8.1 running as a firewall/gateway with PF.
I have a nodephone service with Internode (who have been absolutely
useless in helping me- they point blank refuse, or they say to open
everything right up to their server; which didn't wok anyway btw).
I have been running endless tests on settings changes, tcpdumps on both
the firewall and asterisk, and hours poring over SIP rfc's. I've only
managed to get a headache...
I have tried following best practices, worst practices, and still nothing
works.
My sip.conf looks like this:
[general]
context = default
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
allow = all
;allow = t140red
textsupport = yes
videosupport = yes
;allow = h263
maxcallbitrate = 384
register => sip-in?<phone
number>:<secret>@sip.internode.on.net/<phone number>
externip = <my static ip>
localnet = <my local subnet>
canreinvite = no
hasvoicemail = no
qualify = yes
nat = no
;rtptimeout = 120
rtpkeepalive = 5
;ignoresdpversion = yes
;directmediapermit = <my local subnet>
[sip-in]
type = peer
host = sip.internode.on.net
context = internode-incoming
;externip = <my static ip>
;domain = internode.on.net,internode-incoming
;fromdomain = sip.internode.on.net
;fromuser = <phone number>
;username = <phone number>
;secret = <secret>
;auth = <phone number>:<secret>@BroadWorks
;insecure = invite,port
;register => <phone number>:<secret>@sip.internode.on.net
;nat = never
qualify = yes
canreinvite = no
;expire = 240
[sip-out]
type = peer
host = sip.internode.on.net
context = internode-outgoing
externip = <my static ip>
;username = <phone number>
fromuser = <phone number>
;fromdomain = internode.on.net
;secret = <secret>
;qualify = yes
canreinvite = no
;auth = <phone number>:<secret>@BroadWorks
;nat = never
;pedantic = yes
;insecure = invite,port
;ignoresdpversion = yes
;compactheaders = yes
As you can see I've tried lots of settings. It registers and peers with
the provider, but no outgoing. The provider can call me though.
In extensions.conf:
[internode-outgoing]
exten => _X.,1,Dial(SIP/${EXTEN}@sip-out)
exten => _X.,n,Answer(2)
exten => _X.,n,Playback(ss-noservice)
With debugging enabled, verbose 9, debug 9:
SIP Debugging enabled
<--- SIP read from UDP:<my ata ip>:5060 --->
INVITE sip:0871271201@<asterisk server> SIP/2.0
Via: SIP/2.0/UDP <my ata ip>:5060;branch=z9hG4bK-78cdde11;rport
From: <my ata cid> <sip:<my ata username>@<asterisk
server>>;tag=600053496208a4a8o1
To: <sip:0871271201@<asterisk server>>
Call-ID: e2895c9d-55b90b64@<my ata ip>
CSeq: 101 INVITE
Max-Forwards: 70
Contact: <my ata cid> <sip:<my ata username>@<my ata ip>:5060>
Expires: 240
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 446
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 5330142 5330142 IN IP4 <my ata ip>
s=-
c=IN IP4 <my ata ip>
t=0 0
m=audio 16436 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/--- (14 headers 20 lines) ---
Sending to <my ata ip>:5060 (no NAT)
Using INVITE request as basis request - e2895c9d-55b90b64@<my ata ip>
Found peer '<my ata username>' for '<my ata username>' from <my ata
ip>:5060
<--- Reliably Transmitting (no NAT) to <my ata ip>:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP <my ata ip>:5060;branch=z9hG4bK-78cdde11;received=<my ata
ip>;rport=5060
From: Skinner's Home <sip:<my ata username>@<asterisk
server>>;tag=600053496208a4a8o1
To: <sip:0871271201@<asterisk server>>;tag=as6957dfb9
Call-ID: e2895c9d-55b90b64@192.168.0.196
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="12eb6973"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'e2895c9d-55b90b64@<my ata ip>' in
6400 ms (Method: INVITE)
<--- SIP read from UDP:<my ata ip>:5060 --->
ACK sip:0871271201@<asterisk server> SIP/2.0
Via: SIP/2.0/UDP <my ata ip>:5060;branch=z9hG4bK-78cdde11;rport
From: <my ata cid> <sip:<my ata username>@<asterisk
server>>;tag=600053496208a4a8o1
To: <sip:0871271201@<asterisk server>>;tag=as6957dfb9
Call-ID: e2895c9d-55b90b64@<my ata ip>
CSeq: 101 ACK
Max-Forwards: 70
Contact: <my ata cid> <sip:<my ata username>@<my ata ip>:5060>
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:<my ata ip>:5060 --->
INVITE sip:0871271201@<asterisk server> SIP/2.0
Via: SIP/2.0/UDP <my ata ip>:5060;branch=z9hG4bK-6035d0b9;rport
From: <my ata cid> <sip:<my ata username>@<asterisk
server>>;tag=600053496208a4a8o1
To: <sip:0871271201@<asterisk server>>
Call-ID: e2895c9d-55b90b64@<my ata ip>
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="<my ata
username>",realm="asterisk",nonce="12eb6973",uri="sip:0871271201@<asterisk
server>",algorithm=MD5,response="aa3d9a1719fee78526adb69c56472ceb"
Contact: <my ata cid> <sip:<my ata username>@<my ata ip>:5060>
Expires: 240
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 446
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 5330142 5330142 IN IP4 <my ata ip>
s=-
c=IN IP4 <my ata ip>
t=0 0
m=audio 16436 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
--- (15 headers 20 lines) ---
Sending to <my ata ip>:5060 (no NAT)
Using INVITE request as basis request - e2895c9d-55b90b64@<my ata ip>
Found peer '<my ata username>' for '<my ata username>' from <my ata
ip>:5060
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format G723 for ID 4
Found audio description format PCMA for ID 8
Found audio description format G729a for ID 18
Found audio description format G726-40 for ID 96
Found audio description format G726-24 for ID 97
Found audio description format G726-16 for ID 98
Found audio description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80030c7fffff
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719),
peer - audio=0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x100d0d
(g723|ulaw|alaw|g726|g729|ilbc|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port <my ata ip>:16436
Peer doesn't provide video
Peer doesn't provide T.140
Looking for 0871271201 in users (domain <asterisk server>)
list_route: hop: <sip:<my ata username>@<my ata ip>:5060>
<--- Transmitting (no NAT) to <my ata ip>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <my ata ip>:5060;branch=z9hG4bK-6035d0b9;received=<my ata
ip>;rport=5060
From: <my ata cid> <sip:<my ata username>@<asterisk
server>>;tag=600053496208a4a8o1
To: <sip:0871271201@<asterisk server>>
Call-ID: e2895c9d-55b90b64@<my ata ip>
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:0871271201@<asterisk server>:5060>
Content-Length: 0
<------------>
-- Executing [0871271201@users:1] Goto("SIP/<my ata
username>-0000015c", "internode-outgoing,0871271201,1") in new stack
-- Goto (internode-outgoing,0871271201,1)
-- Executing [0871271201@internode-outgoing:1] Dial("SIP/<my ata
username>-0000015c", "SIP/0871271201@sip-out") in new stack
We think we can do text
And we have a text rtp object
Audio is at 5060
Video is at <my static ip>:5060
Lets set up the text sdp
Text is at <my static 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 <my static ip>:5060;branch=z9hG4bK0c95373b
Max-Forwards: 70
From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
To: <sip:0871271...@sip.internode.on.net>
Contact: <sip:<phone number>@<my static ip>:5060>
Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.1.1
Date: Thu, 10 Feb 2011 02:04:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 738
v=0
o=root 51098296 51098296 IN IP4 <my static ip>
s=Asterisk PBX 1.8.1.1
c=IN IP4 <my static ip>
b=CT:384
t=0 0
m=audio 19850 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/ -- Called 0871271201@sip-out
<--- SIP read from UDP:203.2.134.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK0c95373b
From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
To: <sip:0871271...@sip.internode.on.net>
Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from UDP:203.2.134.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK0c95373b
From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
To: <sip:0871271...@sip.internode.on.net>;tag=232999791-1297303507574
Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
CSeq: 102 INVITE
WWW-Authenticate: DIGEST
qop="auth",nonce="BroadWorksXgjz10gvqTmcdnmtBW",realm="BroadWorks",algorithm=MD5
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 203.2.134.1:5060:
ACK sip:0871271...@sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK0c95373b
Max-Forwards: 70
From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
To: <sip:0871271...@sip.internode.on.net>;tag=232999791-1297303507574
Contact: <sip:<phone number>@<my static ip>:5060>
Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.1.1
Content-Length: 0
---
We think we can do text
And we have a text rtp object
Audio is at 5060
Video is at <my static ip>:5060
Lets set up the text sdp
Text is at <my static 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 <my static ip>:5060;branch=z9hG4bK7634f6ce
Max-Forwards: 70
From: "<my ata username>" <sip:<phone number>@<my static
ip>>;tag=as2c865f34
To: <sip:0871271...@sip.internode.on.net>
Contact: <sip:<phone number>@<my static ip>:5060>
Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.1.1
Authorization: Digest username="<phone number>", realm="BroadWorks",
algorithm=MD5, uri="sip:0871271...@sip.internode.on.net",
nonce="BroadWorksXgjz10gvqTmcdnmtBW",
response="829397204c8dd43c8fc3435baa253075", qop=auth, cnonce="03a77924",
nc=00000001
Date: Thu, 10 Feb 2011 02:04:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 738
v=0
o=root 51098296 51098297 IN IP4 <my static ip>
s=Asterisk PBX 1.8.1.1
c=IN IP4 <my static ip>
b=CT:384
Retransmitting #1 (no NAT) to 203.2.134.1:5060:
INVITE sip:0871271...@sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK7634f6ce
Max-Forwards: 70
From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
To: <sip:0871271...@sip.internode.on.net>
Contact: <sip:0731292848@<my static ip>:5060>
Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.1.1
Authorization: Digest username="<phone number>", realm="BroadWorks",
algorithm=MD5, uri="sip:0871271...@sip.internode.on.net",
nonce="BroadWorksXgjz10gvqTmcdnmtBW",
response="829397204c8dd43c8fc3435baa253075", qop=auth, cnonce="03a77924",
nc=00000001
Date: Thu, 10 Feb 2011 02:04:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 738
v=0
o=root 51098296 51098297 IN IP4 <my static ip>
s=Asterisk PBX 1.8.1.1
c=IN IP4 <my static ip>
b=CT:384
t=0 0Retransmitting #2 (no NAT) to 203.2.134.1:5060:
INVITE sip:0871271...@sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK7634f6ce
Max-Forwards: 70
From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
To: <sip:0871271...@sip.internode.on.net>
Contact: <sip:<phone number>@<my static ip>:5060>
Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.1.1
Authorization: Digest username="<phone number>", realm="BroadWorks",
algorithm=MD5, uri="sip:0871271...@sip.internode.on.net",
nonce="BroadWorksXgjz10gvqTmcdnmtBW",
response="829397204c8dd43c8fc3435baa253075", qop=auth, cnonce="03a77924",
nc=00000001
Date: Thu, 10 Feb 2011 02:04:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 738
v=0
o=root 51098296 51098297 IN IP4 <my static ip>
s=Asterisk PBX 1.8.1.1
c=IN IP4 <my static ip>
b=CT:384
t=0 0Retransmitting #3 (no NAT) to 203.2.134.1:5060:
INVITE sip:0871271...@sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK7634f6ce
Max-Forwards: 70
From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
To: <sip:0871271...@sip.internode.on.net>
Contact: <sip:<phone number>@<my static ip>:5060>
Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.1.1
Authorization: Digest username="<phone number>", realm="BroadWorks",
algorithm=MD5, uri="sip:0871271...@sip.internode.on.net",
nonce="BroadWorksXgjz10gvqTmcdnmtBW",
response="829397204c8dd43c8fc3435baa253075", qop=auth, cnonce="03a77924",
nc=00000001
Date: Thu, 10 Feb 2011 02:04:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 738
v=0
o=root 51098296 51098297 IN IP4 <my static ip>
s=Asterisk PBX 1.8.1.1
c=IN IP4 <my static ip>
b=CT:384
t=0 0Retransmitting #4 (no NAT) to 203.2.134.1:5060:
INVITE sip:0871271...@sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK7634f6ce
Max-Forwards: 70
From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
To: <sip:0871271...@sip.internode.on.net>
Contact: <sip:<phone number>@<my static ip>:5060>
Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.1.1
Authorization: Digest username="<phone number>", realm="BroadWorks",
algorithm=MD5, uri="sip:0871271...@sip.internode.on.net",
nonce="BroadWorksXgjz10gvqTmcdnmtBW",
response="829397204c8dd43c8fc3435baa253075", qop=auth, cnonce="03a77924",
nc=00000001
Date: Thu, 10 Feb 2011 02:04:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 738
v=0
o=root 51098296 51098297 IN IP4 <my static ip>
s=Asterisk PBX 1.8.1.1
c=IN IP4 <my static ip>
b=CT:384
t=0 0Retransmitting #5 (no NAT) to 203.2.134.1:5060:
INVITE sip:0871271...@sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK7634f6ce
Max-Forwards: 70
From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
To: <sip:0871271...@sip.internode.on.net>
Contact: <sip:<phone number>@<my static ip>:5060>
Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.1.1
Authorization: Digest username="<phone number>", realm="BroadWorks",
algorithm=MD5, uri="sip:0871271...@sip.internode.on.net",
nonce="BroadWorksXgjz10gvqTmcdnmtBW",
response="829397204c8dd43c8fc3435baa253075", qop=auth, cnonce="03a77924",
nc=00000001
Date: Thu, 10 Feb 2011 02:04:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 738
v=0
o=root 51098296 51098297 IN IP4 <my static ip>
s=Asterisk PBX 1.8.1.1
c=IN IP4 <my static ip>
b=CT:384
t=0 0Retransmitting #6 (no NAT) to 203.2.134.1:5060:
INVITE sip:0871271...@sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP <my static ip>:5060;branch=z9hG4bK7634f6ce
Max-Forwards: 70
From: "<my ata cid>" <sip:<phone number>@<my static ip>>;tag=as2c865f34
To: <sip:0871271...@sip.internode.on.net>
Contact: <sip:<phone number>@<my static ip>:5060>
Call-ID: 784523d570058f2f64315e506a79ee0f@<my static ip>:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.1.1
Authorization: Digest username="<phone number>", realm="BroadWorks",
algorithm=MD5, uri="sip:0871271...@sip.internode.on.net",
nonce="BroadWorksXgjz10gvqTmcdnmtBW",
response="829397204c8dd43c8fc3435baa253075", qop=auth, cnonce="03a77924",
nc=00000001
Date: Thu, 10 Feb 2011 02:04:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 738
v=0
o=root 51098296 51098297 IN IP4 <my static ip>
s=Asterisk PBX 1.8.1.1
c=IN IP4 <my static ip>
b=CT:384
t=0 0[Feb 10 12:04:20] WARNING[993]: chan_sip.c:3386 retrans_pkt:
Retransmission timeout reached on transmission
784523d570058f2f64315e506a79ee0f@<my static ip>:5060 for seqno 103
(Critical Request) -- See doc/sip-retransmit.txt.
Packet timed out after 6400ms with no response
[Feb 10 12:04:20] WARNING[993]: chan_sip.c:3415 retrans_pkt: Hanging up
call 784523d570058f2f64315e506a79ee0f@<my static ip>:5060 - no reply to
our critical packet (see doc/sip-retransmit.txt).
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [0871271201@internode-outgoing:2] Answer("SIP/<my ata
username>-0000015c", "2") in new stack
Audio is at 5060
Adding codec 0x1 (g723) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding video codec 0x100000 (h263p) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to <my ata ip>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <my ata ip>:5060;branch=z9hG4bK-6035d0b9;received=<my ata
ip>;rport=5060
From: <my ata cid> <sip:<my ata username>@<asterisk
server>>;tag=600053496208a4a8o1
To: <sip:0871271201@<asterisk server>>;tag=as6eeddb0e
Call-ID: e2895c9d-55b90b64@<my ata ip>
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:0871271201@<asterisk server>:5060>
Content-Type: application/sdp
Content-Length: 423
v=0
o=root 1590196377 1590196377 IN IP4 <asterisk server>
s=Asterisk PBX 1.8.1.1
c=IN IP4 <asterisk server>
t=0 0
m=audio 10024 RTP/AVP 4 0 8 18 97 2 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fReally destroying SIP dialog '784523d570058f2f64315e506a79ee0f@<my
static ip>:5060' Method: INVITE
<--- SIP read from UDP:<my ata ip>:5060 --->
ACK sip:0871271201@<asterisk server>:5060 SIP/2.0
Via: SIP/2.0/UDP <my ata ip>:5060;branch=z9hG4bK-3b9bc888;rport
From: <my ata cid> <sip:<my ata username>@<asterisk
server>>;tag=600053496208a4a8o1
To: <sip:0871271201@<asterisk server>>;tag=as6eeddb0e
Call-ID: e2895c9d-55b90b64@<my ata ip>
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="<my ata
username>",realm="asterisk",nonce="12eb6973",uri="sip:0871271201@<asterisk
server>:5060",algorithm=MD5,response="c09a8c20894f257a63225f68d9ef54b7"
Contact: <my ata cid> <sip:<my ata username>@<my ata ip>:5060>
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
-- Executing [0871271201@internode-outgoing:3] Playback("SIP/<my ata
username>-0000015c", "ss-noservice") in new stack
-- <SIP/<my ata usename>-0000015c> Playing 'ss-noservice.gsm'
(language 'en')
<--- SIP read from UDP:<my ata ip>:5060 --->
BYE sip:0871271201@<asterisk server>:5060 SIP/2.0
Via: SIP/2.0/UDP <my ata ip>:5060;branch=z9hG4bK-16d61cfc;rport
From: <my ata cid> <sip:<my ata username>@<asterisk
server>>;tag=600053496208a4a8o1
To: <sip:0871271201@<asterisk server>>;tag=as6eeddb0e
Call-ID: e2895c9d-55b90b64@<my ata ip>
CSeq: 103 BYE
Max-Forwards: 70
Authorization: Digest username="<my ata
username>",realm="asterisk",nonce="12eb6973",uri="sip:0871271201@<asterisk
server>:5060",algorithm=MD5,response="bcad36f00cb422a4e856dec00d73e0d1"
User-Agent: Linksys/PAP2T-3.1.15(LS)
P-RTP-Stat:
PS=502,OS=40160,PR=226,OR=36160,PL=0,JI=0,LA=0,DU=5,EN=G711u,DE=G711u
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to <my ata ip>:5060 (no NAT)
Scheduling destruction of SIP dialog 'e2895c9d-55b90b64@<my ata ip>' in
6400 ms (Method: BYE)
<--- Transmitting (no NAT) to <my ata ip>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <my ata ip>:5060;branch=z9hG4bK-16d61cfc;received=<my ata
ip>;rport=5060
From: <my ata cid> <sip:<my ata username>@<asterisk
server>>;tag=600053496208a4a8o1
To: <sip:0871271201@<asterisk server>>;tag=as6eeddb0e
Call-ID: e2895c9d-55b90b64@<my ata ip>
CSeq: 103 BYE
Server: Asterisk PBX 1.8.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (internode-outgoing, 0871271201, 3) exited non-zero
on 'SIP/<my ata username>-0000015c'
Names and identities have been masked to protect the innocent.
My firewall is setup to binat between asterisk and the static ip, and
failing that my internal network (or dmz, which the asterisk is a part of)
is allowed outgoing traffic natted to the internet.
I've opened up port 5060 and 10000:20000 to the outside world _only_ to
the asterisk server, and the same outgoing.
As near as I can tell asterisk simply can't auth with Internode for some
weird reason. The tcpdumps show 401 from internode, and later a 408-
sometimes. Or just a 408.
The ata could connect, and tcpdumps show invite, 100, 401, then an invite
with auth, then 100, 180, and finally 200 and a conversation.
According to internode they've changed the way it works by turning a peer
to peer connection into a client server model. But I don't think asterisk
is going to play that game.
I _really_ need to see some light of day here. I am new to asterisk, but
I've been playing with firewalls for sometime now. A hint, a clue, a
solution- anything- would be helpful right about now.
TIA
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