[asterisk-users] Error: Invalid SIP message - rejected , no call id
Steven Stromer
filter at stevenstromer.com
Mon Jul 20 18:18:05 CDT 2009
On about 25% of inbound calls to a ring group, picking up any one
extension as it rings results in dead air.
Some details regarding my VoIP network to make the following logs more
readable:
192.168.7.130 resolves to the trixbox host.
192.168.7.135 resolves to endpoint 812.
192.168.7.137 resolves to endpoint 811.
192.168.7.138 resolves to endpoint 810.
192.168.7.139 resolves to endpoint 813.
192.168.7.140 resolves to endpoint 817.
24.136.116.102 is the address of the pbx.
66.23.129.253 is the address of my VoIP provider's peering host.
Very verbose asterisk logging of such a failed inbound call returns
snippets such as the following two examples:
<------------->
[Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 0: ACK sip:18502296800 at phonehome.admiralenvelope.com
SIP/2.0 (57)
[Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 1: Via: SIP/2.0/UDP
192.168.7.140:5060;branch=z9hG4bK8ef20feeb668f72f (66)
[Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 2: From: "Shipping" <sip:817 at phonehome.admiralenvelope.com
>;tag=4aeafc6270620b72 (77)
[Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 3: To: <sip:18502296800 at phonehome.admiralenvelope.com
>;tag=as7823cf0c (66)
[Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 4: Contact: <sip:817 at 192.168.7.140:5060;transport=udp
> (51)
[Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 5: Supported: path (15)
[Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 6: Proxy-
Authorization: Digest username="817", realm="asterisk", algorithm=MD5,
uri="sip:18502296800 at phonehome.admiralenvelope.com", nonce="12f646df",
response="e77e7b202fc6a0bc5930460db8243292" (191)
[Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 7: Call-ID: 9dd235bb45bb93f2 at 192.168.7.140
(39)
[Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 8: CSeq: 61074 ACK (15)
[Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 9: User-Agent:
Grandstream GXP2000 1.1.6.16 (40)
[Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 10: Max-Forwards: 70
(16)
[Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 11: Allow:
INVITE
,ACK
,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
(85)
[Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 12: Content-Length: 0
(17)
[Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Header 13: (0)
[Jul 16 16:17:38] VERBOSE[3214] logger.c: --- (13 headers 0 lines) ---
[Jul 16 16:17:38] DEBUG[3214] chan_sip.c: Invalid SIP message -
rejected , no callid, len 763
[Jul 16 16:17:42] VERBOSE[3214] logger.c:
<--- SIP read from 192.168.7.135:5060 --->
<------------->
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 0: SIP/2.0 200 OK (14)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 1: Via: SIP/2.0/UDP
192.168.7.130:5060;branch=z9hG4bK4ddb9288;rport (64)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 2: From: "Sales: (14)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 3: To: <sip:811 at 192.168.7.137:5060;transport=udp
>;tag=5d9dbfef4e870100 (67)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 4: Call-ID: 238f32201de94e3336a339d650b71349 at 192.168.7.130
(55)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 5: CSeq: 102 INVITE
(16)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 6: User-Agent:
Grandstream GXP2000 1.1.6.16 (40)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 7: Contact: <sip:811 at 192.168.7.137:5060;transport=udp
> (51)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 8: Allow:
INVITE
,ACK
,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
(85)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 9: Content-Type:
application/sdp (29)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 10: Supported:
replaces, timer (26)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 11: Content-Length:
212 (19)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Header 12: (0)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: v=0 (3)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: o=811 8002 8000 IN IP4
192.168.7.137 (36)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: s=SIP Call (10)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: c=IN IP4 192.168.7.137
(22)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: t=0 0 (5)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: m=audio 5008 RTP/AVP 0
101 (26)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: a=sendrecv (10)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: a=rtpmap:0 PCMU/8000
(20)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: a=ptime:20 (10)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: a=rtpmap:101 telephone-
event/8000 (33)
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Line: a=fmtp:101 0-11 (15)
[Jul 16 13:43:42] VERBOSE[3214] logger.c: --- (12 headers 11 lines) ---
[Jul 16 13:43:42] DEBUG[3214] chan_sip.c: Invalid SIP message -
rejected , no callid, len 766
[Jul 16 13:43:42] VERBOSE[3214] logger.c:
<--- SIP read from 192.168.7.135:5060 --->
I performed a tcpdump of UDP packets during one of these failed
inbound calls. Of approx. 3000 packets logged, almost all packets are
a repeat of the following two lines:
67.16.97.188 192.168.7.130 UDP Source port: 53752 Destination port:
19240
192.168.7.130 192.168.7.137 UDP Source port: 16036 Destination port:
wsm-server
A few other logged packets that all seem to be normal registration
traffic between trixbox host and endpoints:
192.168.7.135 192.168.7.130 SIP Status: 200 OK[Packet size limited
during capture]
192.168.7.137 192.168.7.130 SIP Status: 200 OK[Packet size limited
during capture]
192.168.7.138 192.168.7.130 SIP Status: 200 OK[Packet size limited
during capture]
192.168.7.139 192.168.7.130 SIP Status: 200 OK[Packet size limited
during capture]
192.168.7.140 192.168.7.130 SIP Status: 200 OK[Packet size limited
during capture]
192.168.7.130 192.168.7.137 UDP Source port: 16037 Destination port:
wsm-server-ssl
192.168.7.135 192.168.7.130 UDP Source port: sip Destination port: sip
192.168.7.130 192.168.7.135 UDP Source port: sip Destination port: sip
192.168.7.137 192.168.7.130 UDP Source port: sip Destination port: sip
192.168.7.130 192.168.7.137 UDP Source port: sip Destination port: sip
192.168.7.138 192.168.7.130 UDP Source port: sip Destination port: sip
192.168.7.130 192.168.7.138 UDP Source port: sip Destination port: sip
192.168.7.139 192.168.7.130 UDP Source port: sip Destination port: sip
192.168.7.130 192.168.7.139 UDP Source port: sip Destination port: sip
192.168.7.140 192.168.7.130 UDP Source port: sip Destination port: sip
192.168.7.130 192.168.7.140 UDP Source port: sip Destination port: sip
And finally, some more interesting packet entries:
192.168.7.130 67.16.97.188 UDP Source port: 19241 Destination port:
53753
192.168.7.130 67.16.97.188 UDP Source port: 19241 Destination port:
53753 [UDP CHECKSUM INCORRECT] (this repeated 8 times)
192.168.7.130 192.168.7.137 SIP Request: ACK sip:811 at 192.168.7.137:5060;transport=udp
The last packet seems the most interesting, so I provide full packet
details (unfortunately: 'Packet size limited during capture'):
192.168.7.130 192.168.7.137 SIP Request: ACK sip:811 at 192.168.7.137:5060;transport=udp
Frame 185 (425 bytes on wire, 96 bytes captured)
Arrival Time: Jul 16, 2009 16:16:42.274572000
[Time delta from previous captured frame: 0.000942000 seconds]
[Time delta from previous displayed frame: 0.000942000 seconds]
[Time since reference or first frame: 1.820952000 seconds]
Frame Number: 185
Frame Length: 425 bytes
Capture Length: 96 bytes
[Frame is marked: False]
[Protocols in frame: eth:ip:udp:sip]
[Coloring Rule Name: UDP]
[Coloring Rule String: udp]
Ethernet II, Src: Dell_6d:d1:fa (00:12:3f:6d:d1:fa), Dst:
Grandstr_14:48:8d (00:0b:82:14:48:8d)
Destination: Grandstr_14:48:8d (00:0b:82:14:48:8d)
Address: Grandstr_14:48:8d (00:0b:82:14:48:8d)
.... ...0 .... .... .... .... = IG bit: Individual address
(unicast)
.... ..0. .... .... .... .... = LG bit: Globally unique
address (factory default)
Source: Dell_6d:d1:fa (00:12:3f:6d:d1:fa)
Address: Dell_6d:d1:fa (00:12:3f:6d:d1:fa)
.... ...0 .... .... .... .... = IG bit: Individual address
(unicast)
.... ..0. .... .... .... .... = LG bit: Globally unique
address (factory default)
Type: IP (0x0800)
Internet Protocol, Src: 192.168.7.130 (192.168.7.130), Dst:
192.168.7.137 (192.168.7.137)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x60 (DSCP 0x18: Class Selector 3;
ECN: 0x00)
0110 00.. = Differentiated Services Codepoint: Class Selector
3 (0x18)
.... ..0. = ECN-Capable Transport (ECT): 0
.... ...0 = ECN-CE: 0
Total Length: 411
Identification: 0x4026 (16422)
Flags: 0x00
0... = Reserved bit: Not set
.0.. = Don't fragment: Not set
..0. = More fragments: Not set
Fragment offset: 0
Time to live: 64
Protocol: UDP (0x11)
Header checksum: 0xa870 [correct]
[Good: True]
[Bad : False]
Source: 192.168.7.130 (192.168.7.130)
Destination: 192.168.7.137 (192.168.7.137)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Source port: sip (5060)
Destination port: sip (5060)
Length: 391
Checksum: 0x91f4
[Good Checksum: False]
[Bad Checksum: False]
Session Initiation Protocol
Request-Line: ACK sip:811 at 192.168.7.137:5060;transport=udp SIP/2.0
Method: ACK
[Packet size limited during capture: SIP truncated]
My VoIP provider has offered their logs of the same call. They
theorized that the problem exists between the trixbox host and the
endpoints, possibly an issue with an unrecognized codec on the part of
the endpoints. However, the endpoints are equipped for G.729a, so I
don't believe that's the issue here. Their logs are as follows:
U 2009/07/16 20:15:19.097706 66.23.129.253:5060 -> 24.136.116.102:5060
INVITE sip:18888106944 at 24.136.116.102 SIP/2.0..
Record-Route: <sip:18888106944 at 66.23.129.253:5060;nat=yes;ftag=SDju4rb01-gK0665046d;lr=on
>..
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKf06e.46f9bd3.0..
Via: SIP/2.0/UDP
67.16.97.188:5060;branch=z9hG4bKfv7a0p204o3hfkg9r100.1..
From: <sip:8502296800 at 67.16.97.188;isup-oli=0;pstn-
params=808481808882>;tag=SDju4rb01-gK0665046d..
To: <sip:18888106944 at 66.23.129.253>..
Call-ID: SDju4rb01-57c8b4208c5ab4509be51993b36f46eb-v3000i1..
CSeq: 17818 INVITE..
Max-Forwards: 16..
Allow:
INVITE
,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS..
Accept: application/sdp, application/isup, application/dtmf,
application/dtmf-relay, multipart/mixed..
Contact: <sip:8502296800 at 67.16.97.188:5060;transport=udp>..
Remote-Party-ID: <sip:8502296800 at 67.16.97.188:5060>;privacy=off..
Supported: timer..
Session-Expires: 64800..
Min-SE: 64800..
Content-Length: 302..
Content-Disposition: session; handling=required..
Content-Type: application/sdp....
v=0..
o=Sonus_UAC 1891 6087 IN IP4 67.16.97.188..
s=SIP Media Capabilities..
c=IN IP4 67.16.97.188..
t=0 0..
m=audio 53752 RTP/AVP 18 0 8 100..
a=rtpmap:18 G729/8000..
a=fmtp:18 annexb=no..
a=rtpmap:0 PCMU/8000..
a=rtpmap:8 PCMA/8000..
a=rtpmap:100 telephone-event/8000..
a=fmtp:100 0-15..
a=sendrecv..
a=ptime:20..
U 2009/07/16 20:15:19.154046 24.136.116.102:5060 -> 66.23.129.253:5060
SIP/2.0 100 Trying..
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKf06e.
46f9bd3.0;received=66.23.129.253..
Via: SIP/2.0/UDP
67.16.97.188:5060;branch=z9hG4bKfv7a0p204o3hfkg9r100.1..
Record-Route: <sip:18888106944 at 66.23.129.253:5060;nat=yes;ftag=SDju4rb01-gK0665046d;lr=on
>..
From: <sip:8502296800 at 67.16.97.188;isup-oli=0;pstn-
params=808481808882>;tag=SDju4rb01-gK0665046d..
To: <sip:18888106944 at 66.23.129.253>..
Call-ID: SDju4rb01-57c8b4208c5ab4509be51993b36f46eb-v3000i1..
CSeq: 17818 INVITE..
User-Agent: Asterisk PBX..
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..
Supported: replaces..
Contact: <sip:18888106944 at 24.136.116.102>..
Content-Length: 0....
U 2009/07/16 20:15:20.095049 24.136.116.102:5060 -> 66.23.129.253:5060
SIP/2.0 180 Ringing..
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKf06e.
46f9bd3.0;received=66.23.129.253..
Via: SIP/2.0/UDP
67.16.97.188:5060;branch=z9hG4bKfv7a0p204o3hfkg9r100.1..
Record-Route: <sip:18888106944 at 66.23.129.253:5060;nat=yes;ftag=SDju4rb01-gK0665046d;lr=on
>..
From: <sip:8502296800 at 67.16.97.188;isup-oli=0;pstn-
params=808481808882>;tag=SDju4rb01-gK0665046d..
To: <sip:18888106944 at 66.23.129.253>;tag=as0fbef785..
Call-ID: SDju4rb01-57c8b4208c5ab4509be51993b36f46eb-v3000i1..
CSeq: 17818 INVITE..
User-Agent: Asterisk PBX..
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..
Supported: replaces..
Contact: <sip:18888106944 at 24.136.116.102>..
Content-Length: 0....
U 2009/07/16 20:15:20.095897 24.136.116.102:5060 -> 66.23.129.253:5060
SIP/2.0 200 OK..
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKf06e.
46f9bd3.0;received=66.23.129.253..
Via: SIP/2.0/UDP
67.16.97.188:5060;branch=z9hG4bKfv7a0p204o3hfkg9r100.1..
Record-Route: <sip:18888106944 at 66.23.129.253:5060;nat=yes;ftag=SDju4rb01-gK0665046d;lr=on
>..
From: <sip:8502296800 at 67.16.97.188;isup-oli=0;pstn-
params=808481808882>;tag=SDju4rb01-gK0665046d..
To: <sip:18888106944 at 66.23.129.253>;tag=as0fbef785..
Call-ID: SDju4rb01-57c8b4208c5ab4509be51993b36f46eb-v3000i1..
CSeq: 17818 INVITE..
User-Agent: Asterisk PBX..
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..
Supported: replaces..
Contact: <sip:18888106944 at 24.136.116.102>..
Content-Type: application/sdp..
Content-Length: 313....
v=0..
o=root 3200 3200 IN IP4 24.136.116.102..
s=session..
c=IN IP4 24.136.116.102..
t=0 0..
m=audio 19240 RTP/AVP 18 0 8 100..
a=rtpmap:18 G729/8000..
a=fmtp:18 annexb=no..
a=rtpmap:0 PCMU/8000..
a=rtpmap:8 PCMA/8000..
a=rtpmap:100 telephone-event/8000..
a=fmtp:100 0-16..
a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..
U 2009/07/16 20:15:20.135129 66.23.129.253:5060 -> 24.136.116.102:5060
ACK sip:18888106944 at 24.136.116.102 SIP/2.0..
Record-Route: <sip:18888106944 at 66.23.129.253:5060;nat=yes;ftag=SDju4rb01-gK0665046d;lr=on
>..
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=0..
Via: SIP/2.0/UDP
67.16.97.188:5060;branch=z9hG4bKjgfasl305gv04koqe241.1..
From: <sip:8502296800 at 67.16.97.188;isup-oli=0;pstn-
params=808481808882>;tag=SDju4rb01-gK0665046d..
To: <sip:18888106944 at 66.23.129.253>;tag=as0fbef785..
Call-ID: SDju4rb01-57c8b4208c5ab4509be51993b36f46eb-v3000i1..
CSeq: 17818 ACK..
Max-Forwards: 16..
Content-Length: 0....
U 2009/07/16 20:15:20.138585 66.23.129.253:5060 -> 24.136.116.102:5060
INVITE sip:18888106944 at 24.136.116.102 SIP/2.0..
Record-Route: <sip:18888106944 at 66.23.129.253:5060;nat=yes;ftag=SDju4rb01-gK0665046d;lr=on
>..
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bKe06e.e80f1b27.0..
Via: SIP/2.0/UDP
67.16.97.188:5060;branch=z9hG4bKjgfasl305gv04koqe241cb000akd2.1..
From: <sip:8502296800 at 67.16.97.188;isup-oli=0;pstn-
params=808481808882>;tag=SDju4rb01-gK0665046d..
To: <sip:18888106944 at 66.23.129.253>;tag=as0fbef785..
Call-ID: SDju4rb01-57c8b4208c5ab4509be51993b36f46eb-v3000i1..
CSeq: 17819 INVITE..
Max-Forwards: 16..
Allow:
INVITE
,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS..
Accept: application/sdp, application/isup, application/dtmf,
application/dtmf-relay, multipart/mixed..
Contact: <sip:8502296800 at 67.16.97.188:5060;transport=udp>..
Supported: timer..
Session-Expires: 64800;refresher=uac..
Min-SE: 64800..
Content-Length: 254..
Content-Disposition: session; handling=required..
Content-Type: application/sdp....
v=0..
o=Sonus_UAC 1891 6088 IN IP4 67.16.97.188..
s=SIP Media Capabilities..
c=IN IP4 67.16.97.188..
t=0 0..
m=audio 53752 RTP/AVP 18 100..
a=rtpmap:18 G729/8000..
a=fmtp:18 annexb=no..
a=rtpmap:100 telephone-event/8000..
a=fmtp:100 0-15..
a=sendrecv..
a=ptime:20..
U 2009/07/16 20:15:20.162110 24.136.116.102:5060 -> 66.23.129.253:5060
SIP/2.0 100 Trying..
Via: SIP/2.0/UDP
66.23.129.253
:5060;branch=z9hG4bKe06e.e80f1b27.0;received=66.23.129.253..
Via: SIP/2.0/UDP
67.16.97.188:5060;branch=z9hG4bKjgfasl305gv04koqe241cb000akd2.1..
Record-Route: <sip:18888106944 at 66.23.129.253:5060;nat=yes;ftag=SDju4rb01-gK0665046d;lr=on
>..
From: <sip:8502296800 at 67.16.97.188;isup-oli=0;pstn-
params=808481808882>;tag=SDju4rb01-gK0665046d..
To: <sip:18888106944 at 66.23.129.253>;tag=as0fbef785..
Call-ID: SDju4rb01-57c8b4208c5ab4509be51993b36f46eb-v3000i1..
CSeq: 17819 INVITE..
User-Agent: Asterisk PBX..
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..
Supported: replaces..
Contact: <sip:18888106944 at 24.136.116.102>..
Content-Length: 0....
U 2009/07/16 20:15:20.169358 24.136.116.102:5060 -> 66.23.129.253:5060
SIP/2.0 200 OK..
Via: SIP/2.0/UDP
66.23.129.253
:5060;branch=z9hG4bKe06e.e80f1b27.0;received=66.23.129.253..
Via: SIP/2.0/UDP
67.16.97.188:5060;branch=z9hG4bKjgfasl305gv04koqe241cb000akd2.1..
Record-Route: <sip:18888106944 at 66.23.129.253:5060;nat=yes;ftag=SDju4rb01-gK0665046d;lr=on
>..
From: <sip:8502296800 at 67.16.97.188;isup-oli=0;pstn-
params=808481808882>;tag=SDju4rb01-gK0665046d..
To: <sip:18888106944 at 66.23.129.253>;tag=as0fbef785..
Call-ID: SDju4rb01-57c8b4208c5ab4509be51993b36f46eb-v3000i1..
CSeq: 17819 INVITE..
User-Agent: Asterisk PBX..
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..
Supported: replaces..
Contact: <sip:18888106944 at 24.136.116.102>..
Content-Type: application/sdp..
Content-Length: 265....
v=0..
o=root 3200 3201 IN IP4 24.136.116.102..
s=session..
c=IN IP4 24.136.116.102..
t=0 0..
m=audio 19240RTP/AVP 18 100..
a=rtpmap:18 G729/8000..
a=fmtp:18 annexb=no..
a=rtpmap:100 telephone-event/8000..
a=fmtp:100 0-16..
a=silenceSupp:off - - - -..
a=ptime:20..
a=sendrecv..
U 2009/07/16 20:15:20.184889 66.23.129.253:5060 -> 24.136.116.102:5060
ACK sip:18888106944 at 24.136.116.102 SIP/2.0..
Record-Route: <sip:18888106944 at 66.23.129.253:5060;nat=yes;ftag=SDju4rb01-gK0665046d;lr=on
>..
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=0..
Via: SIP/2.0/UDP
67.16.97.188:5060;branch=z9hG4bKlo9lkp0068jgkl8ei5s1.1..
From: <sip:8502296800 at 67.16.97.188;isup-oli=0;pstn-
params=808481808882>;tag=SDju4rb01-gK0665046d..
To: <sip:18888106944 at 66.23.129.253>;tag=as0fbef785..
Call-ID: SDju4rb01-57c8b4208c5ab4509be51993b36f46eb-v3000i1..
CSeq: 17819 ACK..
Max-Forwards: 16..
Content-Length: 0....
U 2009/07/16 20:17:17.560838 66.23.129.253:5060 -> 24.136.116.102:5060
BYE sip:18888106944 at 24.136.116.102 SIP/2.0..
Record-Route: <sip:18888106944 at 66.23.129.253:5060;nat=yes;ftag=SDju4rb01-gK0665046d;lr=on
>..
Via: SIP/2.0/UDP 66.23.129.253:5060;branch=z9hG4bK306e.d826f45.0..
Via: SIP/2.0/UDP
67.16.97.188:5060;branch=z9hG4bKjgfasl305gv04koqe241cd000a4e2.1..
From: <sip:8502296800 at 67.16.97.188;isup-oli=0;pstn-
params=808481808882>;tag=SDju4rb01-gK0665046d..
To: <sip:18888106944 at 66.23.129.253>;tag=as0fbef785..
Call-ID: SDju4rb01-57c8b4208c5ab4509be51993b36f46eb-v3000i1..
CSeq: 17820 BYE..
Max-Forwards: 16..
Content-Length: 0....
U 2009/07/16 20:17:17.581096 24.136.116.102:5060 -> 66.23.129.253:5060
SIP/2.0 200 OK..
Via: SIP/2.0/UDP
66.23.129.253:5060;branch=z9hG4bK306e.d826f45.0;received=66.23.129.253..
Via: SIP/2.0/UDP
67.16.97.188:5060;branch=z9hG4bKjgfasl305gv04koqe241cd000a4e2.1..
Record-Route: <sip:18888106944 at 66.23.129.253:5060;nat=yes;ftag=SDju4rb01-gK0665046d;lr=on
>..
From: <sip:8502296800 at 67.16.97.188;isup-oli=0;pstn-
params=808481808882>;tag=SDju4rb01-gK0665046d..
To: <sip:18888106944 at 66.23.129.253>;tag=as0fbef785..
Call-ID: SDju4rb01-57c8b4208c5ab4509be51993b36f46eb-v3000i1..
CSeq: 17820 BYE..
User-Agent: Asterisk PBX..
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..
Supported: replaces..
Contact: <sip:18888106944 at 24.136.116.102>..
Content-Length: 0....
I recognize that I am posting to an asterisk list, but I think this is
a question best asked of an asterisk/SIP proficient group. I hope you
will all be soft on me there! My trixbox version is v2.6.2.1. I am
aware that there are updates available, but this is a production
system, and I generally try to not fix what isn't surely broke. On the
other hand, if an upgrade will resolve this issue (as in, 'it's a
known bug'), I will happily do so!
All endpoints are Grandstream GXP2000s.
On a final note, my custom iptables did not report anything being
blocked during this period.
If anyone can advise, it would be much appreciated. Thanks in advance!
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