[asterisk-users] Unable to make outgoing calls with Internode

Dovid Bender asteriskusers at dovid.net
Wed Feb 9 22:00:20 CST 2011


Hi,

Under sip-out why do you have secret, fromdomain and NAT commented out ?

Also it seems like Asterisk is re-transmitting which means it seems like it 
is not getting any response from your ISP. It could be a firewall issue, it 
could be your ISP. If your ISP refuses to work with you you may want to go 
with an ISP that will help.

Regards,

Dovid

----- Original Message ----- 
From: "Da Rock" <asterisk-users at herveybayaustralia.com.au>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Thursday, February 10, 2011 05:08
Subject: [asterisk-users] Unable to make outgoing calls with Internode


> 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 at 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 at users:1] Goto("SIP/<my ata 
> username>-0000015c", "internode-outgoing,0871271201,1") in new stack
>     -- Goto (internode-outgoing,0871271201,1)
>     -- Executing [0871271201 at internode-outgoing:1] Dial("SIP/<my ata 
> username>-0000015c", "SIP/0871271201 at 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:0871271201 at 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:0871271201 at 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 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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:0871271201 at 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 at 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 at 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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