On Thu, Oct 23, 2008 at 12:39 PM, Juan Rodríguez <[EMAIL PROTECTED]> wrote:
> And this phone are connected in a local LAN??
> Because I see Asterisk receiving a "Bad request" from 68.156.63.118
> If those phones are not in your local LAN, try with a soft phone first.
> Could be Zoiper or Xlite.
> Besides, use SIP SET DEBUG, for SIP debugging and try to see why is 101
> sending a "400 Bad request" back to Asterisk.
>
Both of these phones are on my local lan but the Asterisk server is at
a colo facility on the internet outside of the local lan. The local
lan does use NAT/PAT. I see an error "Warning: 399 Bad Request -
'Malformed/Missing FROM: field'. Is this a problem?
Thanks
---
ns1*CLI>
<--- SIP read from 68.156.63.118:1082 --->
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14
From: <sip:[EMAIL PROTECTED];user=phone>;tag=2678814914
To: <sip:[EMAIL PROTECTED];user=phone>
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED]:1083;user=phone;transport=udp>
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Expires: 300
Content-Length: 274
Content-Type: application/sdp
v=0
o=102 157742 157742 IN IP4 172.16.2.18
s=Cisco 7912 SIP Call
c=IN IP4 68.156.63.118
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (14 headers 12 lines) ---
Sending to 68.156.63.118 : 1083 (no NAT)
Using INVITE request as basis request - [EMAIL PROTECTED]
<--- Reliably Transmitting (NAT) to 68.156.63.118:1082 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14;received=68.156.63.118
From: <sip:[EMAIL PROTECTED];user=phone>;tag=2678814914
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as355e0f84
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="ns1.neocipher.net",
nonce="7c2e1ba9"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000
ms (Method: INVITE)
Found user '102'
<--- SIP read from 64.2.142.116:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060
From: <sip:[EMAIL PROTECTED]>;tag=as401a34d4
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 3064 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from 64.2.142.116:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK68f8877d;received=209.251.157.91;rport=5060
From: <sip:[EMAIL PROTECTED]>;tag=as401a34d4
To: <sip:[EMAIL PROTECTED]>;tag=as7a2f92a1
Call-ID: [EMAIL PROTECTED]
CSeq: 3064 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="575628ec"
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Responding to challenge, registration to domain/host name inbound18.vitelity.net
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 64.2.142.116:5060:
REGISTER sip:inbound18.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK6245e988;rport
From: <sip:[EMAIL PROTECTED]>;tag=as751cb0af
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 3065 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="rsreese", realm="asterisk",
algorithm=MD5, uri="sip:inbound18.vitelity.net", nonce="575628ec",
response="b765dbdebba8af18b19707efe651d65d"
Expires: 120
Contact: <sip:[EMAIL PROTECTED]>
Event: registration
Content-Length: 0
---
<--- SIP read from 68.156.63.118:1082 --->
ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bK5a88bbfa3bc85d14
From: <sip:[EMAIL PROTECTED];user=phone>;tag=2678814914
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as355e0f84
Call-ID: [EMAIL PROTECTED]
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: Cisco-CP7912/8.0.1-060412A
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
ns1*CLI>
<--- SIP read from 68.156.63.118:1082 --->
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bKd82d2d0850902de2
From: <sip:[EMAIL PROTECTED];user=phone>;tag=2678814914
To: <sip:[EMAIL PROTECTED];user=phone>
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED]:1083;user=phone;transport=udp>
User-Agent: Cisco-CP7912/8.0.1-060412A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces, 100rel
Proxy-Authorization: Digest
username="102",realm="ns1.neocipher.net",nonce="7c2e1ba9",uri="sip:[EMAIL
PROTECTED]",response="105dfec593cbcfac83380461870c3a07"
Expires: 300
Content-Length: 274
Content-Type: application/sdp
v=0
o=102 157750 157750 IN IP4 172.16.2.18
s=Cisco 7912 SIP Call
c=IN IP4 68.156.63.118
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (15 headers 12 lines) ---
Sending to 68.156.63.118 : 1082 (NAT)
Using INVITE request as basis request - [EMAIL PROTECTED]
Found user '102'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 68.156.63.118:16384
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 68.156.63.118:16384
Looking for 101 in default (domain neocipher.net)
list_route: hop: <sip:[EMAIL PROTECTED]:1083;user=phone;transport=udp>
<--- Transmitting (NAT) to 68.156.63.118:1082 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
68.156.63.118:1083;branch=z9hG4bKd82d2d0850902de2;received=68.156.63.118
From: <sip:[EMAIL PROTECTED];user=phone>;tag=2678814914
To: <sip:[EMAIL PROTECTED];user=phone>
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
<------------>
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/102-081e4968",
"SIP/101&SIP/[EMAIL PROTECTED]|30") in new stack
Audio is at 209.251.157.91 port 10532
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 68.156.63.118:1038:
INVITE sip:[EMAIL PROTECTED]:1039;transport=udp SIP/2.0
Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK7db3f711;rport
From: "Stephen\" <sip:[EMAIL PROTECTED]>;tag=as5b299b78
To: <sip:[EMAIL PROTECTED]:1039;transport=udp>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 23 Oct 2008 17:39:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 5235 5235 IN IP4 209.251.157.91
s=session
c=IN IP4 209.251.157.91
t=0 0
m=audio 10532 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 101
Audio is at 209.251.157.91 port 18610
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 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.2.142.17:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK1b46ea9b;rport
From: "Stephen\" <sip:[EMAIL PROTECTED]>;tag=as0f273e60
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Stephen\" <sip:[EMAIL PROTECTED]>;privacy=off;screen=no
Date: Thu, 23 Oct 2008 17:39:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 428
v=0
o=root 5235 5235 IN IP4 209.251.157.91
s=session
c=IN IP4 209.251.157.91
t=0 0
m=audio 18610 RTP/AVP 0 3 8 112 5 10 7 111 101
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:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called [EMAIL PROTECTED]
ns1*CLI>
<--- SIP read from 64.2.142.116:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK6245e988;received=209.251.157.91;rport=5060
From: <sip:[EMAIL PROTECTED]>;tag=as751cb0af
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 3065 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
ns1*CLI>
<--- SIP read from 64.2.142.116:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK6245e988;received=209.251.157.91;rport=5060
From: <sip:[EMAIL PROTECTED]>;tag=as751cb0af
To: <sip:[EMAIL PROTECTED]>;tag=as7a2f92a1
Call-ID: [EMAIL PROTECTED]
CSeq: 3065 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 60
Contact: <sip:[EMAIL PROTECTED]>;expires=60
Date: Thu, 23 Oct 2008 17:27:25 GMT
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Scheduling destruction of SIP dialog
'[EMAIL PROTECTED]' in 32000 ms (Method:
REGISTER)
[Oct 23 13:39:52] NOTICE[5264]: chan_sip.c:12682
handle_response_register: Outbound Registration: Expiry for
inbound18.vitelity.net is 60 sec (Scheduling reregistration in 45 s)
<--- SIP read from 64.2.142.17:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
209.251.157.91:5060;branch=z9hG4bK1b46ea9b;received=209.251.157.91;rport=5060
From: "Stephen\" <sip:[EMAIL PROTECTED]>;tag=as0f273e60
To: <sip:[EMAIL PROTECTED]>;tag=as0d5d9875
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 64.2.142.17:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK1b46ea9b;rport
From: "Stephen\" <sip:[EMAIL PROTECTED]>;tag=as0f273e60
To: <sip:[EMAIL PROTECTED]>;tag=as0d5d9875
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Stephen\" <sip:[EMAIL PROTECTED]>;privacy=off;screen=no
Content-Length: 0
---
-- SIP/vitel-outbound-081f9240 is circuit-busy
ns1*CLI>
<--- SIP read from 68.156.63.118:1038 --->
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK7db3f711;rport
From: "Stephen\" <sip:[EMAIL PROTECTED]>;tag=as5b299b78
To: <sip:[EMAIL PROTECTED]:1039;transport=udp>
Call-ID: [EMAIL PROTECTED]
Warning: 399 Bad Request - 'Malformed/Missing FROM: field'
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
-- Got SIP response 400 "Bad Request" back from 68.156.63.118
Transmitting (NAT) to 68.156.63.118:1038:
ACK sip:[EMAIL PROTECTED]:1039;transport=udp SIP/2.0
Via: SIP/2.0/UDP 209.251.157.91:5060;branch=z9hG4bK7db3f711;rport
From: "Stephen\" <sip:[EMAIL PROTECTED]>;tag=as5b299b78
To: <sip:[EMAIL PROTECTED]:1039;transport=udp>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Really destroying SIP dialog
'[EMAIL PROTECTED]' Method: INVITE
-- SIP/101-08195e68 is circuit-busy
== Everyone is busy/congested at this time (2:0/2/0)
-- Executing [EMAIL PROTECTED]:2] GotoIf("SIP/102-081e4968",
"0?lbl_default_1:") in new stack
-- Executing [EMAIL PROTECTED]:3] GotoIf("SIP/102-081e4968",
"0?lbl_default_1:") in new stack
-- Executing [EMAIL PROTECTED]:4] Hangup("SIP/102-081e4968", "") in new
stack
== Spawn extension (default, 101, 4) exited non-zero on 'SIP/102-081e4968'
Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000
ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 68.156.63.118:1082 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP
68.156.63.118:1083;branch=z9hG4bKd82d2d0850902de2;received=68.156.63.118
From: <sip:[EMAIL PROTECTED];user=phone>;tag=2678814914
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as74c3be83
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
<------------>
Really destroying SIP dialog
'[EMAIL PROTECTED]' Method: INVITE
ns1*CLI>
<--- SIP read from 68.156.63.118:1082 --->
ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0
Via: SIP/2.0/UDP 68.156.63.118:1083;branch=z9hG4bKd82d2d0850902de2
From: <sip:[EMAIL PROTECTED];user=phone>;tag=2678814914
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as74c3be83
Call-ID: [EMAIL PROTECTED]
CSeq: 2 ACK
Max-Forwards: 70
User-Agent: Cisco-CP7912/8.0.1-060412A
Proxy-Authorization: Digest
username="102",realm="ns1.neocipher.net",nonce="7c2e1ba9",uri="sip:[EMAIL
PROTECTED]",response="105dfec593cbcfac83380461870c3a07"
Content-Length: 0
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