thr3ads.net - asterisk users - [asterisk-users] Call Dropped in Voicemail / No Reply to Our Critical Packet w/ SIP Debug [Jan 2009]

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Lincoln King-Cliby
2009-Jan-20 19:07 UTC
[asterisk-users] Call Dropped in Voicemail / No Reply to Our Critical Packet w/ SIP Debug

Hi All,

A long time ago I posted about an issue where calls on one of our Asterisk boxes
were being dropped in Voicemail (and 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).".

I thought the issue had cleared itself up, but it has resurfaced. We're
running 1.4.22 on built from source, using Cisco 7961G phones with the SIP
firmware image (and I've tried most of the recent firmware versions for the
Cisco phones to see if that would change things at all).

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


<--- Reliably Transmitting (no NAT) to 10.2.0.203:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK1971acea;received=10.2.0.203
From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906943a6bb290-9435a462
To: <sip:Voicemail at 10.2.0.2>;tag=as00aa5042
Call-ID: 001d45b6-1d490087-44d5cce8-d599695a at 10.2.0.203
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="1db91b00"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '001d45b6-1d490087-44d5cce8-d599695a 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 - 001d45b6-1d490087-44d5cce8-d599695a 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 10.2.0.203:29422
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 10.2.0.203:29422
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=z9hG4bKddc0a0b8;received=10.2.0.203
From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906943a6bb290-9435a462
To: <sip:Voicemail at 10.2.0.2>
Call-ID: 001d45b6-1d490087-44d5cce8-d599695a 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 12088
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=z9hG4bKddc0a0b8;received=10.2.0.203
From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906943a6bb290-9435a462
To: <sip:Voicemail at 10.2.0.2>;tag=as58c8eec4
Call-ID: 001d45b6-1d490087-44d5cce8-d599695a 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 12088 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 14:33:01] NOTICE[15644]: sched.c:220 ast_sched_add_variable: Scheduled
event in 0 ms?
Sending to 10.2.0.203 : 5060 (no NAT)
cworks-phones1*CLI>
<--- 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=z9hG4bK8a220b9f;received=10.2.0.203
From: "Jim Felderman" <sip:1101 at
10.2.0.2>;tag=001d45b61d4906943a6bb290-9435a462
To: <sip:Voicemail at 10.2.0.2>;tag=as58c8eec4
Call-ID: 001d45b6-1d490087-44d5cce8-d599695a at 10.2.0.203
CSeq: 103 BYE
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


<------------>
Really destroying SIP dialog '001d45b6-1d490087-44d5cce8-d599695a at
10.2.0.203' Method: BYE
Sending to 10.2.0.203 : 5060 (no NAT)
cworks-phones1*CLI>
<--- Reliably Transmitting (no NAT) to 10.2.0.203:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK51921801;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=as537c53ce
Call-ID: 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="20c9330e"
Content-Length: 0


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