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Lincoln King-Cliby

2009-Feb-02 17:39 UTC

[asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail

Hi All, 

I posted this a couple weeks ago with no response, I'm hoping that someone
will see it this time around and be so kind as to offer advice for resolving
this issue (or point me in the direction of a better place to ask)

"Some" (but not all) calls on one of our Asterisk boxes are being
dropped in Voicemail -- only in voicemail -- after about 20 seconds with the
error logged "[Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1980
retrans_pkt: Hanging up call 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203 -
no reply to our critical packet (see doc/sip-retransmit.txt).".

We're running Asterisk 1.4.22 built from source and Cisco 7961G phones with
the SIP firmware image. I've tried most of the recent firmware versions for
the phones with no real impact on the issue. Strange thing is that while all of
the phones use a variation on the same config file (with the only changes being
the SIP account details and speed dial keys) but one user in particular seems to
suffer the issue far more frequently.

I would appreciate any assistance since I'm stumped. The output of SIP DEBUG
for the extension most frequently affected by the issue is below; starting with
one call to voicemail that was successfully completed, followed by a 2nd call
that was dropped after approximately 18 seconds.

The issue is consistently inconsistent - it doesn't happen on every call to
Voicemail, but those that it does happen on it's always within the first
approximately 20 seconds of the call; once you pass the 25 second mark
you're free and clear for that call-it will not be dropped. It also seems
like it's possible to reproduce the issue by making several calls to
Voicemail in short order, but this isn't the only trigger as sometimes the
first call to voicemail in 12+ hours will also trigger it.

I'm also baffled by the fact that this ONLY crops up on calls to Voicemail
on the local box; SIP to SIP calls on the same Asterisk box, SIP to SIP calls
from this Asterisk box to an Asterisk Appliance at a remote site, SIP to POTS,
and POTS to SIP calls are completely unaffected.

Again, any advice/suggestions/things to look at/etc are greatly appreciated! 

Thanks in advance, 

Lincoln

<------------>
Scheduling destruction of SIP dialog '001d45b6-1d490088-46ef7ed6-adbeb875 at
10.2.0.203' in 32000 ms (Method: INVITE) Sending to 10.2.0.203 : 5060 (no
NAT) Using INVITE request as basis request - 001d45b6-1d490088-46ef7ed6-adbeb875
at 10.2.0.203
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.2.0.203:24394 Found audio description format PCMU
for ID 0 Found audio description format PCMA for ID 8 Found audio description
format G729 for ID 18 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 10.2.0.203:24394
Looking for Voicemail in internal (domain 10.2.0.2)
list_route: hop: <sip:1101 at 10.2.0.203:5060;transport=udp>
cworks-phones1*CLI>
<--- Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
To: <sip:Voicemail at 10.2.0.2>
Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:Voicemail at 10.2.0.2>
Content-Length: 0


<------------>
Audio is at 10.2.0.2 port 13256
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:Voicemail at 10.2.0.2>
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 27452 27452 IN IP4 10.2.0.2
s=session
c=IN IP4 10.2.0.2
t=0 0
m=audio 13256 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/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

<------------>
Retransmitting #1 (no NAT) to 10.2.0.203:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:Voicemail at 10.2.0.2>
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 27452 27452 IN IP4 10.2.0.2
s=session
c=IN IP4 10.2.0.2
t=0 0
m=audio 13256 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/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

---
Retransmitting #2 (no NAT) to 10.2.0.203:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:Voicemail at 10.2.0.2>
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 27452 27452 IN IP4 10.2.0.2
s=session
c=IN IP4 10.2.0.2
t=0 0
m=audio 13256 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/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

---
Retransmitting #3 (no NAT) to 10.2.0.203:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:Voicemail at 10.2.0.2>
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 27452 27452 IN IP4 10.2.0.2
s=session
c=IN IP4 10.2.0.2
t=0 0
m=audio 13256 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/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

---
Scheduling destruction of SIP dialog '44b7d5c43fff7c0567e6c3be3d7d69c9 at
10.2.0.2' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to
10.2.0.203:5060:
NOTIFY sip:1101 at 10.2.0.203:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.2.0.2:5060;branch=z9hG4bK59292f64;rport
From: "asterisk" <sip:asterisk at 10.2.0.2>;tag=as73ca9f87
To: <sip:1101 at 10.2.0.203:5060;transport=udp>
Contact: <sip:asterisk at 10.2.0.2>
Call-ID: 44b7d5c43fff7c0567e6c3be3d7d69c9 at 10.2.0.2
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 83

Messages-Waiting: yes
Message-Account: sip:asterisk at 10.2.0.2
Voice-Message: 3/5

---
Really destroying SIP dialog '44b7d5c43fff7c0567e6c3be3d7d69c9 at
10.2.0.2' Method: NOTIFY Retransmitting #4 (no NAT) to 10.2.0.203:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:Voicemail at 10.2.0.2>
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 27452 27452 IN IP4 10.2.0.2
s=session
c=IN IP4 10.2.0.2
t=0 0
m=audio 13256 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/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

---
Retransmitting #5 (no NAT) to 10.2.0.203:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:Voicemail at 10.2.0.2>
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 27452 27452 IN IP4 10.2.0.2
s=session
c=IN IP4 10.2.0.2
t=0 0
m=audio 13256 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/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

---
Retransmitting #6 (no NAT) to 10.2.0.203:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:Voicemail at 10.2.0.2>
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 27452 27452 IN IP4 10.2.0.2
s=session
c=IN IP4 10.2.0.2
t=0 0
m=audio 13256 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/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

---
[Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1958 retrans_pkt: Maximum retries
exceeded on transmission 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203 for
seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1980 retrans_pkt: Hanging up call
001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203 - no reply to our critical
packet (see doc/sip-retransmit.txt).
Really destroying SIP dialog '001d45b6-1d490088-46ef7ed6-adbeb875 at
10.2.0.203' Method: INVITE Sending to 10.2.0.203 : 5060 (no NAT)
cworks-phones1*CLI>
<--- Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 481 Call
leg/transaction does not exist
Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK446b7de3;received=10.2.0.203
From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '0d32ee8515d9f6dc4439002f1d6016a5 at
10.2.0.2' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to
10.2.0.203:5060:
NOTIFY sip:1101 at 10.2.0.203:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.2.0.2:5060;branch=z9hG4bK0cb71f67;rport
From: "asterisk" <sip:asterisk at 10.2.0.2>;tag=as0b88d5a9
To: <sip:1101 at 10.2.0.203:5060;transport=udp>
Contact: <sip:asterisk at 10.2.0.2>
Call-ID: 0d32ee8515d9f6dc4439002f1d6016a5 at 10.2.0.2
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 83

Messages-Waiting: yes
Message-Account: sip:asterisk at 10.2.0.2
Voice-Message: 2/6

---
Really destroying SIP dialog '0d32ee8515d9f6dc4439002f1d6016a5 at
10.2.0.2' Method: NOTIFY cworks-phones1*CLI>


--
Lincoln King-Cliby, CTS
Applications Engineer
ControlWorks Consulting, LLC
V: 440-729-4640 x1107?F: 440-729-0884 I: http://www.thecontrolworks.com/
Crestron Authorized Independent Programmer

Steve Totaro

2009-Feb-02 17:52 UTC

head link

[asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail

On Mon, Feb 2, 2009 at 12:39 PM, Lincoln King-Cliby
<lincoln at controlworks.com> wrote:> Hi All,
>
> I posted this a couple weeks ago with no response, I'm hoping that
someone will see it this time around and be so kind as to offer advice for
resolving this issue (or point me in the direction of a better place to ask)
>
> "Some" (but not all) calls on one of our Asterisk boxes are being
dropped in Voicemail -- only in voicemail -- after about 20 seconds with the
error logged "[Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1980
retrans_pkt: Hanging up call 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203 -
no reply to our critical packet (see doc/sip-retransmit.txt).".
>
> We're running Asterisk 1.4.22 built from source and Cisco 7961G phones
with the SIP firmware image. I've tried most of the recent firmware versions
for the phones with no real impact on the issue. Strange thing is that while all
of the phones use a variation on the same config file (with the only changes
being the SIP account details and speed dial keys) but one user in particular
seems to suffer the issue far more frequently.
>
> I would appreciate any assistance since I'm stumped. The output of SIP
DEBUG for the extension most frequently affected by the issue is below; starting
with one call to voicemail that was successfully completed, followed by a 2nd
call that was dropped after approximately 18 seconds.
>
> The issue is consistently inconsistent - it doesn't happen on every
call to Voicemail, but those that it does happen on it's always within the
first approximately 20 seconds of the call; once you pass the 25 second mark
you're free and clear for that call-it will not be dropped. It also seems
like it's possible to reproduce the issue by making several calls to
Voicemail in short order, but this isn't the only trigger as sometimes the
first call to voicemail in 12+ hours will also trigger it.
>
> I'm also baffled by the fact that this ONLY crops up on calls to
Voicemail on the local box; SIP to SIP calls on the same Asterisk box, SIP to
SIP calls from this Asterisk box to an Asterisk Appliance at a remote site, SIP
to POTS, and POTS to SIP calls are completely unaffected.
>
> Again, any advice/suggestions/things to look at/etc are greatly
appreciated!
>
> Thanks in advance,
>
> Lincoln
>
> <------------>
> Scheduling destruction of SIP dialog
'001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203' in 32000 ms (Method:
INVITE) Sending to 10.2.0.203 : 5060 (no NAT) Using INVITE request as basis
request - 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 18
> Found RTP audio format 101
> Peer audio RTP is at port 10.2.0.203:24394 Found audio description format
PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio
description format G729 for ID 18 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 10.2.0.203:24394
Looking for Voicemail in internal (domain 10.2.0.2)
> list_route: hop: <sip:1101 at 10.2.0.203:5060;transport=udp>
> cworks-phones1*CLI>
> <--- Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Length: 0
>
>
> <------------>
> Audio is at 10.2.0.2 port 13256
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
>
> <--- Reliably Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0
200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
>
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
>
> <------------>
> Retransmitting #1 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
>
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
>
> ---
> Retransmitting #2 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
>
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
>
> ---
> Retransmitting #3 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
>
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
>
> ---
> Scheduling destruction of SIP dialog '44b7d5c43fff7c0567e6c3be3d7d69c9
at 10.2.0.2' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to
10.2.0.203:5060:
> NOTIFY sip:1101 at 10.2.0.203:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 10.2.0.2:5060;branch=z9hG4bK59292f64;rport
> From: "asterisk" <sip:asterisk at 10.2.0.2>;tag=as73ca9f87
> To: <sip:1101 at 10.2.0.203:5060;transport=udp>
> Contact: <sip:asterisk at 10.2.0.2>
> Call-ID: 44b7d5c43fff7c0567e6c3be3d7d69c9 at 10.2.0.2
> CSeq: 102 NOTIFY
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Event: message-summary
> Content-Type: application/simple-message-summary
> Content-Length: 83
>
> Messages-Waiting: yes
> Message-Account: sip:asterisk at 10.2.0.2
> Voice-Message: 3/5
>
> ---
> Really destroying SIP dialog '44b7d5c43fff7c0567e6c3be3d7d69c9 at
10.2.0.2' Method: NOTIFY Retransmitting #4 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
>
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
>
> ---
> Retransmitting #5 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
>
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
>
> ---
> Retransmitting #6 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
>
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
>
> ---
> [Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1958 retrans_pkt: Maximum
retries exceeded on transmission 001d45b6-1d490088-46ef7ed6-adbeb875 at
10.2.0.203 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
> [Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1980 retrans_pkt: Hanging up
call 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203 - no reply to our
critical packet (see doc/sip-retransmit.txt).
> Really destroying SIP dialog '001d45b6-1d490088-46ef7ed6-adbeb875 at
10.2.0.203' Method: INVITE Sending to 10.2.0.203 : 5060 (no NAT)
cworks-phones1*CLI>
> <--- Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 481 Call
leg/transaction does not exist
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK446b7de3;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 103 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog '0d32ee8515d9f6dc4439002f1d6016a5
at 10.2.0.2' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to
10.2.0.203:5060:
> NOTIFY sip:1101 at 10.2.0.203:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 10.2.0.2:5060;branch=z9hG4bK0cb71f67;rport
> From: "asterisk" <sip:asterisk at 10.2.0.2>;tag=as0b88d5a9
> To: <sip:1101 at 10.2.0.203:5060;transport=udp>
> Contact: <sip:asterisk at 10.2.0.2>
> Call-ID: 0d32ee8515d9f6dc4439002f1d6016a5 at 10.2.0.2
> CSeq: 102 NOTIFY
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Event: message-summary
> Content-Type: application/simple-message-summary
> Content-Length: 83
>
> Messages-Waiting: yes
> Message-Account: sip:asterisk at 10.2.0.2
> Voice-Message: 2/6
>
> ---
> Really destroying SIP dialog '0d32ee8515d9f6dc4439002f1d6016a5 at
10.2.0.2' Method: NOTIFY cworks-phones1*CLI>
>
>
> --
> Lincoln King-Cliby, CTS
> Applications Engineer
> ControlWorks Consulting, LLC
> V: 440-729-4640 x1107 F: 440-729-0884 I: http://www.thecontrolworks.com/
Crestron Authorized Independent Programmer
>
>
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
I have a customer with the same complaint and I am trying to figure it
out as well.  I have not caught the debug action yet though.

First, are you using FreePBX?  Second, are you using the "announce"
feature.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

David Gibbons

2009-Feb-02 17:55 UTC

head link

[asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail

Which firmware load? We had all kinds of trouble with 8.4.x, after being stable
for a few months on 8.3.x. Going back to 8.3.x made all of the weirdness
disappear. While we're on the cisco note, I have  script to remotely reboot
the SIP firmware load Ciscos and to provision the phones based on active
directory if you're interested... back on topic:

Have you run a packet cap on a mirror of the switchport the phone this is
happening on is connected to? Anything strange? What's happening on the
switch backplane (network backbone) at large when you notice the problems? Major
transfers/lots of traffic? Anything else running on the * server?

--Dave

<snip>
We're running Asterisk 1.4.22 built from source and Cisco 7961G phones with
the SIP firmware image. I've tried most of the recent firmware versions for
the phones with no real impact on the issue. Strange thing is that while all of
the phones use a variation on the same config file (with the only changes being
the SIP account details and speed dial keys) but one user in particular seems to
suffer the issue far more frequently.
</snip>

Alex Balashov

2009-Feb-02 17:59 UTC

head link

[asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail

Sounds like there's some sort of firewall in place or something else 
that is preventing an ACK from being received in response to the 200 OK. 
  Notice that the 200 OK keeps being retransmitted.

Lincoln King-Cliby wrote:
> Hi All, 
> 
> I posted this a couple weeks ago with no response, I'm hoping that
someone will see it this time around and be so kind as to offer advice for
resolving this issue (or point me in the direction of a better place to ask)
> 
> "Some" (but not all) calls on one of our Asterisk boxes are being
dropped in Voicemail -- only in voicemail -- after about 20 seconds with the
error logged "[Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1980
retrans_pkt: Hanging up call 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203 -
no reply to our critical packet (see doc/sip-retransmit.txt).".
> 
> We're running Asterisk 1.4.22 built from source and Cisco 7961G phones
with the SIP firmware image. I've tried most of the recent firmware versions
for the phones with no real impact on the issue. Strange thing is that while all
of the phones use a variation on the same config file (with the only changes
being the SIP account details and speed dial keys) but one user in particular
seems to suffer the issue far more frequently.
> 
> I would appreciate any assistance since I'm stumped. The output of SIP
DEBUG for the extension most frequently affected by the issue is below; starting
with one call to voicemail that was successfully completed, followed by a 2nd
call that was dropped after approximately 18 seconds.
> 
> The issue is consistently inconsistent - it doesn't happen on every
call to Voicemail, but those that it does happen on it's always within the
first approximately 20 seconds of the call; once you pass the 25 second mark
you're free and clear for that call-it will not be dropped. It also seems
like it's possible to reproduce the issue by making several calls to
Voicemail in short order, but this isn't the only trigger as sometimes the
first call to voicemail in 12+ hours will also trigger it.
> 
> I'm also baffled by the fact that this ONLY crops up on calls to
Voicemail on the local box; SIP to SIP calls on the same Asterisk box, SIP to
SIP calls from this Asterisk box to an Asterisk Appliance at a remote site, SIP
to POTS, and POTS to SIP calls are completely unaffected.
> 
> Again, any advice/suggestions/things to look at/etc are greatly
appreciated!
> 
> Thanks in advance, 
> 
> Lincoln
> 
> <------------>
> Scheduling destruction of SIP dialog
'001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203' in 32000 ms (Method:
INVITE) Sending to 10.2.0.203 : 5060 (no NAT) Using INVITE request as basis
request - 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 18
> Found RTP audio format 101
> Peer audio RTP is at port 10.2.0.203:24394 Found audio description format
PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio
description format G729 for ID 18 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 10.2.0.203:24394
Looking for Voicemail in internal (domain 10.2.0.2)
> list_route: hop: <sip:1101 at 10.2.0.203:5060;transport=udp>
> cworks-phones1*CLI>
> <--- Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Length: 0
> 
> 
> <------------>
> Audio is at 10.2.0.2 port 13256
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> 
> <--- Reliably Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0
200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
> 
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
> 
> <------------>
> Retransmitting #1 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
> 
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
> 
> ---
> Retransmitting #2 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
> 
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
> 
> ---
> Retransmitting #3 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
> 
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
> 
> ---
> Scheduling destruction of SIP dialog '44b7d5c43fff7c0567e6c3be3d7d69c9
at 10.2.0.2' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to
10.2.0.203:5060:
> NOTIFY sip:1101 at 10.2.0.203:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 10.2.0.2:5060;branch=z9hG4bK59292f64;rport
> From: "asterisk" <sip:asterisk at 10.2.0.2>;tag=as73ca9f87
> To: <sip:1101 at 10.2.0.203:5060;transport=udp>
> Contact: <sip:asterisk at 10.2.0.2>
> Call-ID: 44b7d5c43fff7c0567e6c3be3d7d69c9 at 10.2.0.2
> CSeq: 102 NOTIFY
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Event: message-summary
> Content-Type: application/simple-message-summary
> Content-Length: 83
> 
> Messages-Waiting: yes
> Message-Account: sip:asterisk at 10.2.0.2
> Voice-Message: 3/5
> 
> ---
> Really destroying SIP dialog '44b7d5c43fff7c0567e6c3be3d7d69c9 at
10.2.0.2' Method: NOTIFY Retransmitting #4 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
> 
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
> 
> ---
> Retransmitting #5 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
> 
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
> 
> ---
> Retransmitting #6 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
> 
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
> 
> ---
> [Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1958 retrans_pkt: Maximum
retries exceeded on transmission 001d45b6-1d490088-46ef7ed6-adbeb875 at
10.2.0.203 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
> [Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1980 retrans_pkt: Hanging up
call 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203 - no reply to our
critical packet (see doc/sip-retransmit.txt).
> Really destroying SIP dialog '001d45b6-1d490088-46ef7ed6-adbeb875 at
10.2.0.203' Method: INVITE Sending to 10.2.0.203 : 5060 (no NAT)
cworks-phones1*CLI>
> <--- Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 481 Call
leg/transaction does not exist
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK446b7de3;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 103 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
> 
> 
> <------------>
> Scheduling destruction of SIP dialog '0d32ee8515d9f6dc4439002f1d6016a5
at 10.2.0.2' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to
10.2.0.203:5060:
> NOTIFY sip:1101 at 10.2.0.203:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 10.2.0.2:5060;branch=z9hG4bK0cb71f67;rport
> From: "asterisk" <sip:asterisk at 10.2.0.2>;tag=as0b88d5a9
> To: <sip:1101 at 10.2.0.203:5060;transport=udp>
> Contact: <sip:asterisk at 10.2.0.2>
> Call-ID: 0d32ee8515d9f6dc4439002f1d6016a5 at 10.2.0.2
> CSeq: 102 NOTIFY
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Event: message-summary
> Content-Type: application/simple-message-summary
> Content-Length: 83
> 
> Messages-Waiting: yes
> Message-Account: sip:asterisk at 10.2.0.2
> Voice-Message: 2/6
> 
> ---
> Really destroying SIP dialog '0d32ee8515d9f6dc4439002f1d6016a5 at
10.2.0.2' Method: NOTIFY cworks-phones1*CLI>
> 
> 
> --
> Lincoln King-Cliby, CTS
> Applications Engineer
> ControlWorks Consulting, LLC
> V: 440-729-4640 x1107 F: 440-729-0884 I: http://www.thecontrolworks.com/
Crestron Authorized Independent Programmer
> 
> 
> 
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

Steve J. Douglas

2009-Feb-03 08:29 UTC

head link

[asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail

Hi Lincoln,

Asterisk was expecting ACK after sending the 200 OK message. After 
repeated attempts at sending the 200 OK message and not receiving ACK, 
it terminated the call. Are you able to do a packet capture on the phone 
end? Mostly likely the phone is sending the ACK, but its either sent to 
somewhere else or your firewall is blocking it (not likely since you are 
able to receive the call in the first place). The packet capture on the 
phone end will probably show you the smoking gun.

Regards,
Steve

Lincoln King-Cliby wrote:> Hi All, 
>
> I posted this a couple weeks ago with no response, I'm hoping that
someone will see it this time around and be so kind as to offer advice for
resolving this issue (or point me in the direction of a better place to ask)
>
> "Some" (but not all) calls on one of our Asterisk boxes are being
dropped in Voicemail -- only in voicemail -- after about 20 seconds with the
error logged "[Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1980
retrans_pkt: Hanging up call 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203 -
no reply to our critical packet (see doc/sip-retransmit.txt).".
>
> We're running Asterisk 1.4.22 built from source and Cisco 7961G phones
with the SIP firmware image. I've tried most of the recent firmware versions
for the phones with no real impact on the issue. Strange thing is that while all
of the phones use a variation on the same config file (with the only changes
being the SIP account details and speed dial keys) but one user in particular
seems to suffer the issue far more frequently.
>
> I would appreciate any assistance since I'm stumped. The output of SIP
DEBUG for the extension most frequently affected by the issue is below; starting
with one call to voicemail that was successfully completed, followed by a 2nd
call that was dropped after approximately 18 seconds.
>
> The issue is consistently inconsistent - it doesn't happen on every
call to Voicemail, but those that it does happen on it's always within the
first approximately 20 seconds of the call; once you pass the 25 second mark
you're free and clear for that call-it will not be dropped. It also seems
like it's possible to reproduce the issue by making several calls to
Voicemail in short order, but this isn't the only trigger as sometimes the
first call to voicemail in 12+ hours will also trigger it.
>
> I'm also baffled by the fact that this ONLY crops up on calls to
Voicemail on the local box; SIP to SIP calls on the same Asterisk box, SIP to
SIP calls from this Asterisk box to an Asterisk Appliance at a remote site, SIP
to POTS, and POTS to SIP calls are completely unaffected.
>
> Again, any advice/suggestions/things to look at/etc are greatly
appreciated!
>
> Thanks in advance, 
>
> Lincoln
>
> <------------>
> Scheduling destruction of SIP dialog
'001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203' in 32000 ms (Method:
INVITE) Sending to 10.2.0.203 : 5060 (no NAT) Using INVITE request as basis
request - 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 18
> Found RTP audio format 101
> Peer audio RTP is at port 10.2.0.203:24394 Found audio description format
PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio
description format G729 for ID 18 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 10.2.0.203:24394
Looking for Voicemail in internal (domain 10.2.0.2)
> list_route: hop: <sip:1101 at 10.2.0.203:5060;transport=udp>
> cworks-phones1*CLI>
> <--- Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Length: 0
>
>
> <------------>
> Audio is at 10.2.0.2 port 13256
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
>
> <--- Reliably Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0
200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
>
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
>
> <------------>
> Retransmitting #1 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
>
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
>
> ---
> Retransmitting #2 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
>
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
>
> ---
> Retransmitting #3 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
>
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
>
> ---
> Scheduling destruction of SIP dialog '44b7d5c43fff7c0567e6c3be3d7d69c9
at 10.2.0.2' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to
10.2.0.203:5060:
> NOTIFY sip:1101 at 10.2.0.203:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 10.2.0.2:5060;branch=z9hG4bK59292f64;rport
> From: "asterisk" <sip:asterisk at 10.2.0.2>;tag=as73ca9f87
> To: <sip:1101 at 10.2.0.203:5060;transport=udp>
> Contact: <sip:asterisk at 10.2.0.2>
> Call-ID: 44b7d5c43fff7c0567e6c3be3d7d69c9 at 10.2.0.2
> CSeq: 102 NOTIFY
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Event: message-summary
> Content-Type: application/simple-message-summary
> Content-Length: 83
>
> Messages-Waiting: yes
> Message-Account: sip:asterisk at 10.2.0.2
> Voice-Message: 3/5
>
> ---
> Really destroying SIP dialog '44b7d5c43fff7c0567e6c3be3d7d69c9 at
10.2.0.2' Method: NOTIFY Retransmitting #4 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
>
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
>
> ---
> Retransmitting #5 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
>
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
>
> ---
> Retransmitting #6 (no NAT) to 10.2.0.203:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:Voicemail at 10.2.0.2>
> Content-Type: application/sdp
> Content-Length: 256
>
> v=0
> o=root 27452 27452 IN IP4 10.2.0.2
> s=session
> c=IN IP4 10.2.0.2
> t=0 0
> m=audio 13256 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/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
>
> ---
> [Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1958 retrans_pkt: Maximum
retries exceeded on transmission 001d45b6-1d490088-46ef7ed6-adbeb875 at
10.2.0.203 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
> [Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1980 retrans_pkt: Hanging up
call 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203 - no reply to our
critical packet (see doc/sip-retransmit.txt).
> Really destroying SIP dialog '001d45b6-1d490088-46ef7ed6-adbeb875 at
10.2.0.203' Method: INVITE Sending to 10.2.0.203 : 5060 (no NAT)
cworks-phones1*CLI>
> <--- Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 481 Call
leg/transaction does not exist
> Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK446b7de3;received=10.2.0.203
> From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b
> To: <sip:Voicemail at 10.2.0.2>;tag=as53449c29
> Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
> CSeq: 103 BYE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog '0d32ee8515d9f6dc4439002f1d6016a5
at 10.2.0.2' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to
10.2.0.203:5060:
> NOTIFY sip:1101 at 10.2.0.203:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 10.2.0.2:5060;branch=z9hG4bK0cb71f67;rport
> From: "asterisk" <sip:asterisk at 10.2.0.2>;tag=as0b88d5a9
> To: <sip:1101 at 10.2.0.203:5060;transport=udp>
> Contact: <sip:asterisk at 10.2.0.2>
> Call-ID: 0d32ee8515d9f6dc4439002f1d6016a5 at 10.2.0.2
> CSeq: 102 NOTIFY
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Event: message-summary
> Content-Type: application/simple-message-summary
> Content-Length: 83
>
> Messages-Waiting: yes
> Message-Account: sip:asterisk at 10.2.0.2
> Voice-Message: 2/6
>
> ---
> Really destroying SIP dialog '0d32ee8515d9f6dc4439002f1d6016a5 at
10.2.0.2' Method: NOTIFY cworks-phones1*CLI>
>
>
> --
> Lincoln King-Cliby, CTS
> Applications Engineer
> ControlWorks Consulting, LLC
> V: 440-729-4640 x1107 F: 440-729-0884 I: http://www.thecontrolworks.com/
Crestron Authorized Independent Programmer
>
>
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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>

Mark Wiater

2009-Feb-03 19:24 UTC

head link

[asterisk-users] "No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail

Lincoln King-Cliby wrote:>> -----Original Message-----
> 
> Then starting at packet 3217 there are a series 6 of ICMP
> "Destination unreachable (Port Unreachable)" messages from the
> Asterisk server to the phone, with an RTP packet from the Phone
> to the Asterisk server before each Destination unreachable
> message.
> 
Wouldn't this suggest that either Asterisk couldn't open the port,
or opened it and then closed it? Or I suppose that perhaps the phone
and asterisk didn't negotiate the port properly?

Does your packet capture show that the phone is consistently using
the correct port to communicate with the * server? There's no change
or anything, right? You don't happen to have a corresponding sip
debug to this wireshark capture do you? You might be able to
correlate info from the two.

In your original post, I thought I read that you could reproduce
this issue by increasing load on the asterisk server.  What does the
caller experience in the first 20 seconds when a call to voicemail
is going to fail? Just ringing?

Any chance there's anything in Asterisk's or the OSes logs about
some failure of the network stack? What OS is this?

Mark

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