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

[Steve Totaro]

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
>
>
>
>
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>

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)