Problem: Asterisk is ignoring INVITE Messages

stani · April 24, 2008, 1:46pm

Hi!

We are experementing with SIP and Asterisk in school.
We had no big problems so far, but now we have a problem with sending an INVITE-Message to another host, because Asterisk ignores them.
And we cannot figure out where this failure comes from.

So, please help us to get this thing done.

Here is the SIP debug:
REGISTER-Part:

<--- SIP read from 192.168.100.17:5070 --->
REGISTER sip:192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.17:5070;branch=z9hG4bK-I0qc
From: <sip:2000@192.168.100.32>;tag=16cD
To: <sip:2000@192.168.100.32>
Call-ID: I0qc@192.168.100.17
CSeq: 2 REGISTER
User-Agent: TwinSIP Communicator
Contact: <sip:2000@192.168.100.17:5070>
Expires: 3600
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.100.17 : 5070 (no NAT)

<--- Transmitting (no NAT) to 192.168.100.17:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.17:5070;branch=z9hG4bK-I0qc;received=192.168.100.17
From: <sip:2000@192.168.100.32>;tag=16cD
To: <sip:2000@192.168.100.32>
Call-ID: I0qc@192.168.100.17
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2000@192.168.100.32>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 192.168.100.17:5070 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.17:5070;branch=z9hG4bK-I0qc;received=192.168.100.17
From: <sip:2000@192.168.100.32>;tag=16cD
To: <sip:2000@192.168.100.32>;tag=as0d8811c3
Call-ID: I0qc@192.168.100.17
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="12105d19"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'I0qc@192.168.100.17' in 32000 ms (Method: REGISTER)

<--- SIP read from 192.168.100.17:5070 --->
REGISTER sip:192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.17:5070;maddr=;ttl=;received=;branch=z9hG4bK6mvpR;rport=
From: <sip:2000@192.168.100.32>;tag=16cD
To: <sip:2000@192.168.100.32>
Call-ID: I0qc@192.168.100.17
CSeq: 2 REGISTER
User-Agent: TwinSIP Communicator
Contact: <sip:2000@192.168.100.17:5070>;q=;expires=
Expires: 3600
Content-Length: 0
Authorization: Digest username="2000", realm="asterisk", nonce="12105d19", uri="192.168.100.32", response="474418e865cc40a226cd6d54820f0bce", algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.100.17 : 5070 (no NAT)

<--- Transmitting (no NAT) to 192.168.100.17:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.17:5070;maddr=;ttl=;received=;branch=z9hG4bK6mvpR;rport=;received=192.168.100.17
From: <sip:2000@192.168.100.32>;tag=16cD
To: <sip:2000@192.168.100.32>
Call-ID: I0qc@192.168.100.17
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2000@192.168.100.32>
Content-Length: 0


<------------>
    -- Saved useragent "TwinSIP Communicator" for peer 2000

<--- Transmitting (no NAT) to 192.168.100.17:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.17:5070;maddr=;ttl=;received=;branch=z9hG4bK6mvpR;rport=;received=192.168.100.17
From: <sip:2000@192.168.100.32>;tag=16cD
To: <sip:2000@192.168.100.32>;tag=as0d8811c3
Call-ID: I0qc@192.168.100.17
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:2000@192.168.100.17:5070>;expires=3600
Date: Thu, 24 Apr 2008 15:29:15 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'I0qc@192.168.100.17' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog 'I0qc@192.168.100.17' Method: REGISTER

<--- SIP read from 192.168.100.50:5070 --->
REGISTER sip:192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-U$Ã–O
From: <sip:2001@192.168.100.32>;tag=YilY
To: <sip:2001@192.168.100.32>
Call-ID: U$Ã–O@192.168.100.50
CSeq: 1 REGISTER
User-Agent: TwinSIP Communicator
Contact: <sip:2001@192.168.100.50:5070>
Expires: 3600
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.100.50 : 5070 (no NAT)

<--- Transmitting (no NAT) to 192.168.100.50:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-U$Ã–O;received=192.168.100.50
From: <sip:2001@192.168.100.32>;tag=YilY
To: <sip:2001@192.168.100.32>
Call-ID: U$Ã–O@192.168.100.50
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2001@192.168.100.32>
Content-Length: 0


<------------>

<--- Transmitting (no NAT) to 192.168.100.50:5070 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-U$Ã–O;received=192.168.100.50
From: <sip:2001@192.168.100.32>;tag=YilY
To: <sip:2001@192.168.100.32>;tag=as77afec68
Call-ID: U$Ã–O@192.168.100.50
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="06c7a8b6"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'U$Ã–O@192.168.100.50' in 32000 ms (Method: REGISTER)

<--- SIP read from 192.168.100.50:5070 --->
REGISTER sip:192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;maddr=;ttl=;received=;branch=z9hG4bKÃŸ$fyT;rport=
From: <sip:2001@192.168.100.32>;tag=YilY
To: <sip:2001@192.168.100.32>
Call-ID: U$Ã–O@192.168.100.50
CSeq: 1 REGISTER
User-Agent: TwinSIP Communicator
Contact: <sip:2001@192.168.100.50:5070>;q=;expires=
Expires: 3600
Content-Length: 0
Authorization: Digest username="2001", realm="asterisk", nonce="06c7a8b6", uri="192.168.100.32", response="0097b6ddf1cec8fc636ccb938dcb0e38", algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.100.50 : 5070 (no NAT)

<--- Transmitting (no NAT) to 192.168.100.50:5070 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.50:5070;maddr=;ttl=;received=;branch=z9hG4bKÃŸ$fyT;rport=;received=192.168.100.50
From: <sip:2001@192.168.100.32>;tag=YilY
To: <sip:2001@192.168.100.32>
Call-ID: U$Ã–O@192.168.100.50
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2001@192.168.100.32>
Content-Length: 0


<------------>
    -- Saved useragent "TwinSIP Communicator" for peer 2001

<--- Transmitting (no NAT) to 192.168.100.50:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.50:5070;maddr=;ttl=;received=;branch=z9hG4bKÃŸ$fyT;rport=;received=192.168.100.50
From: <sip:2001@192.168.100.32>;tag=YilY
To: <sip:2001@192.168.100.32>;tag=as77afec68
Call-ID: U$Ã–O@192.168.100.50
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:2001@192.168.100.50:5070>;expires=3600
Date: Thu, 24 Apr 2008 15:29:51 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'U$Ã–O@192.168.100.50' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog 'U$Ã–O@192.168.100.50' Method: REGISTER

INVITE-Part:

<— SIP read from 192.168.100.50:5070 —>
INVITE sip:2000@192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: sip:2001@192.168.100.32
To: sip:2000@192.168.100.32
Call-ID: eE6h@192.168.100.32
CSeq: 3 INVITE
Contact: sip:2001@192.168.100.50
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99

v=0
o=andy 2634 2634 IN IP4 192.168.100.50
s=test von alex
t=0 0
m=audio 9050 udp PCM/32000/8/1

<------------->
— (10 headers 5 lines) —
Sending to 192.168.100.50 : 5070 (no NAT)
Using INVITE request as basis request - eE6h@192.168.100.32

<— Reliably Transmitting (no NAT) to 192.168.100.50:5070 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50
From: sip:2001@192.168.100.32
To: sip:2000@192.168.100.32;tag=as3887b859
Call-ID: eE6h@192.168.100.32
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4e5f93f7"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘eE6h@192.168.100.32’ in 32000 ms (Method: INVITE)
Found user ‘2001’

<— SIP read from 192.168.100.50:5070 —>
INVITE sip:2000@192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: sip:2001@192.168.100.32
To: sip:2000@192.168.100.32;tag=as3887b859
Call-ID: eE6h@192.168.100.32
CSeq: 3 INVITE
Contact: sip:2001@192.168.100.50
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username=“2001”, realm=“asterisk”, nonce=“4e5f93f7”, uri=“192.168.100.32”, response=“39abd8d49d07c1cbb489b0fc467d2856”, algorithm=MD5

<------------->
— (11 headers 0 lines) —
[size=150][color=red]Ignoring this INVITE request[/color][/size]
Retransmitting #1 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50
From: sip:2001@192.168.100.32
To: sip:2000@192.168.100.32;tag=as3887b859
Call-ID: eE6h@192.168.100.32
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4e5f93f7"
Content-Length: 0

<— SIP read from 192.168.100.50:5070 —>
INVITE sip:2000@192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: sip:2001@192.168.100.32
To: sip:2000@192.168.100.32;tag=as3887b859
Call-ID: eE6h@192.168.100.32
CSeq: 3 INVITE
Contact: sip:2001@192.168.100.50
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username=“2001”, realm=“asterisk”, nonce=“4e5f93f7”, uri=“192.168.100.32”, response=“39abd8d49d07c1cbb489b0fc467d2856”, algorithm=MD5

<------------->
— (11 headers 0 lines) —
[size=150][color=red]Ignoring this INVITE request[/color][/size]
Retransmitting #2 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50
From: sip:2001@192.168.100.32
To: sip:2000@192.168.100.32;tag=as3887b859
Call-ID: eE6h@192.168.100.32
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4e5f93f7"
Content-Length: 0

<— SIP read from 192.168.100.50:5070 —>
INVITE sip:2000@192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: sip:2001@192.168.100.32
To: sip:2000@192.168.100.32;tag=as3887b859
Call-ID: eE6h@192.168.100.32
CSeq: 3 INVITE
Contact: sip:2001@192.168.100.50
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username=“2001”, realm=“asterisk”, nonce=“4e5f93f7”, uri=“192.168.100.32”, response=“39abd8d49d07c1cbb489b0fc467d2856”, algorithm=MD5

<------------->
— (11 headers 0 lines) —
[size=150][color=red]Ignoring this INVITE request[/color][/size]
Retransmitting #3 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50
From: sip:2001@192.168.100.32
To: sip:2000@192.168.100.32;tag=as3887b859
Call-ID: eE6h@192.168.100.32
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4e5f93f7"
Content-Length: 0

<— SIP read from 192.168.100.50:5070 —>
INVITE sip:2000@192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: sip:2001@192.168.100.32
To: sip:2000@192.168.100.32;tag=as3887b859
Call-ID: eE6h@192.168.100.32
CSeq: 3 INVITE
Contact: sip:2001@192.168.100.50
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username=“2001”, realm=“asterisk”, nonce=“4e5f93f7”, uri=“192.168.100.32”, response=“39abd8d49d07c1cbb489b0fc467d2856”, algorithm=MD5

<------------->
— (11 headers 0 lines) —
[size=150][color=red]Ignoring this INVITE request[/color][/size]
Retransmitting #4 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50
From: sip:2001@192.168.100.32
To: sip:2000@192.168.100.32;tag=as3887b859
Call-ID: eE6h@192.168.100.32
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4e5f93f7"
Content-Length: 0

<— SIP read from 192.168.100.50:5070 —>
INVITE sip:2000@192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: sip:2001@192.168.100.32
To: sip:2000@192.168.100.32;tag=as3887b859
Call-ID: eE6h@192.168.100.32
CSeq: 3 INVITE
Contact: sip:2001@192.168.100.50
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username=“2001”, realm=“asterisk”, nonce=“4e5f93f7”, uri=“192.168.100.32”, response=“39abd8d49d07c1cbb489b0fc467d2856”, algorithm=MD5

<------------->
— (11 headers 0 lines) —
[size=150][color=red]Ignoring this INVITE request[/color][/size]
Retransmitting #5 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50
From: sip:2001@192.168.100.32
To: sip:2000@192.168.100.32;tag=as3887b859
Call-ID: eE6h@192.168.100.32
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4e5f93f7"
Content-Length: 0

<— SIP read from 192.168.100.50:5070 —>
INVITE sip:2000@192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: sip:2001@192.168.100.32
To: sip:2000@192.168.100.32;tag=as3887b859
Call-ID: eE6h@192.168.100.32
CSeq: 3 INVITE
Contact: sip:2001@192.168.100.50
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username=“2001”, realm=“asterisk”, nonce=“4e5f93f7”, uri=“192.168.100.32”, response=“39abd8d49d07c1cbb489b0fc467d2856”, algorithm=MD5

<------------->
— (11 headers 0 lines) —
[size=150][color=red]Ignoring this INVITE request[/color][/size]
Retransmitting #6 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50
From: sip:2001@192.168.100.32
To: sip:2000@192.168.100.32;tag=as3887b859
Call-ID: eE6h@192.168.100.32
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4e5f93f7"
Content-Length: 0

<— SIP read from 192.168.100.50:5070 —>
INVITE sip:2000@192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: sip:2001@192.168.100.32
To: sip:2000@192.168.100.32;tag=as3887b859
Call-ID: eE6h@192.168.100.32
CSeq: 3 INVITE
Contact: sip:2001@192.168.100.50
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username=“2001”, realm=“asterisk”, nonce=“4e5f93f7”, uri=“192.168.100.32”, response=“39abd8d49d07c1cbb489b0fc467d2856”, algorithm=MD5

<------------->
— (11 headers 0 lines) —
[size=150][color=red]Ignoring this INVITE request[/color][/size]
[Apr 24 17:31:57] WARNING[5391]: chan_sip.c:1939 retrans_pkt: Maximum retries exceeded on transmission eE6h@192.168.100.32 for seqno 3 (Non-critical Response)
Really destroying SIP dialog ‘eE6h@192.168.100.32’ Method: INVITE

End of INVITE-Part.

Is anything wrong with the SIP-Messages? Or what can we do to solve this problem?

Thanks in advance,
Alex