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Intermittent incomplete calls - 418 Call/Transaction Does Not Exist


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We're using hosted snom ONE ver 4.0.1.3499 (Linux). We're running into an intermittent issue where when you dial a number and hit the check mark it grabs a shared line you'll hear the comfort noise and then the light goes out and the call goes dead. Now the call does still go through to the other side but the callee gets dead air. We've tried 3 different providers, and 2 different Internet connections (Comcast and Windstream T1), as well as 3 different routers (Linksys WRT54GL, PFSense, SonicWall TZ180). The phones are Ssnom 320s and 370s. I've only seen it happen when we dial the number with the receiver on the hook, hit the check mark, and then pick up the phone before it starts to ring. However, I've had clients call with this problem that state they weren't using speakerphone. Sometimes if you hit the redial button, it'll do it again, and sometimes it'll go right through.

 

Here is the call flow in the logfile:

 

x's are the last 4 of callee's phone number

 

[7] 2011/08/17 10:55:36:

SIP Rx tls:64.139.77.137:54696:

INVITE sip:xxxxxxx@pbx.allstatecomputers.com;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-5sw0ts03k0wc;rport
From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg
To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone>
Call-ID: 3c4b75252475-v0yf8yotze2n
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:405@192.168.100.120:1402;transport=tls;line=5z91mad1>;reg-id=1
X-Serialnumber: 00041326601A
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom370/8.2.29
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Require: buttons
Content-Type: application/sdp
Content-Length: 530


[7] 2011/08/17 10:55:36:

SIP Tx tls:64.139.77.137:54696:

SIP/2.0 100 Trying
Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-5sw0ts03k0wc;rport=54696;received=64.139.77.137
From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg
To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone>;tag=aa112c7956
Call-ID: 3c4b75252475-v0yf8yotze2n
CSeq: 1 INVITE
Content-Length: 0


[7] 2011/08/17 10:55:36:

Set packet length to 20


[6] 2011/08/17 10:55:36:

Sending RTP for 3c4b75252475-v0yf8yotze2n#aa112c7956 to 192.168.100.120:59716


[7] 2011/08/17 10:55:36:

SIP Tx udp:208.115.60.142:5060:


INVITE sip:1561427xxx@208.115.60.142;user=phone SIP/2.0
Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-3e3e98944e760e7f42026e1128561203;rport
From: <sip:5617431521@208.115.60.142>;tag=174310263
To: <sip:1561427xxxx@208.115.60.142;user=phone>
Call-ID: a086938b@pbx
CSeq: 14029 INVITE
Max-Forwards: 70
Contact: <sip:5617431521@208.115.60.136:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: AllstateTelecom-PBX/4.0.1.3499
Content-Type: application/sdp
Content-Length: 290

v=0
o=- 1066957623 1066957623 IN IP4 208.115.60.136
s=-
c=IN IP4 208.115.60.136
t=0 0
m=audio 61400 RTP/AVP 18 0 101
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv




[7] 2011/08/17 10:55:36:

Set packet length to 20



[6] 2011/08/17 10:55:36:

Send codec pcmu/8000



[7] 2011/08/17 10:55:36:

SIP Tx tls:64.139.77.137:54696:



SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-5sw0ts03k0wc;rport=54696;received=64.139.77.137
From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg
To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone>;tag=aa112c7956
Call-ID: 3c4b75252475-v0yf8yotze2n
CSeq: 1 INVITE
Contact: <sip:405@208.115.60.136:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: AllstateTelecom-PBX/4.0.1.3499
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 484

v=0
o=- 1439637230 1439637230 IN IP4 208.115.60.136
s=-
c=IN IP4 208.115.60.136
t=0 0
m=audio 54918 RTP/AVP 0 8 9 18 2 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:kdbRvVLTrWAjCHLIafsjJnz9s8mbK+JTtKFDOscV
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv


[7] 2011/08/17 10:55:36:

SIP Rx udp:208.115.60.142:5060:

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-3e3e98944e760e7f42026e1128561203;rport=5060
From: <sip:5617431521@208.115.60.142>;tag=174310263
To: <sip:1561427xxxx@208.115.60.142;user=phone>;tag=7696ca4181be7f5787517527c78a3092.7f39
Call-ID: a086938b@pbx
CSeq: 14029 INVITE
Proxy-Authenticate: Digest realm="208.115.60.142", nonce="4e4bd777000004bf7965b8ab80081eb702aeab3b0d6c916b"
Server: OpenSIPS (1.6.4-2-tls (i386/linux))
Content-Length: 0


[7] 2011/08/17 10:55:36:

SIP Tx udp:208.115.60.142:5060:

ACK sip:1561427xxxx@208.115.60.142;user=phone SIP/2.0
Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-3e3e98944e760e7f42026e1128561203;rport
From: <sip:5617431521@208.115.60.142>;tag=174310263
To: <sip:1561427xxxx@208.115.60.142;user=phone>;tag=7696ca4181be7f5787517527c78a3092.7f39
Call-ID: a086938b@pbx
CSeq: 14029 ACK
Max-Forwards: 70
Content-Length: 0


[7] 2011/08/17 10:55:36:

SIP Tx udp:208.115.60.142:5060:

INVITE sip:1561427xxxx@208.115.60.142;user=phone SIP/2.0
Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-5e0d32504503372356c54f08ea6ef395;rport
From: <sip:5617431521@208.115.60.142>;tag=174310263
To: <sip:1561427xxxx@208.115.60.142;user=phone>
Call-ID: a086938b@pbx
CSeq: 14030 INVITE
Max-Forwards: 70
Contact: <sip:5617431521@208.115.60.136:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: AllstateTelecom-PBX/4.0.1.3499
Proxy-Authorization: Digest realm="208.115.60.142",nonce="4e4bd777000004bf7965b8ab80081eb702aeab3b0d6c916b",response="643e10a3e747539004c22ef2b9175779",username="5617431521",uri="sip:1561427xxxx@208.115.60.142;user=phone",algorithm=MD5
Content-Type: application/sdp
Content-Length: 290

v=0
o=- 1066957623 1066957623 IN IP4 208.115.60.136
s=-
c=IN IP4 208.115.60.136
t=0 0
m=audio 61400 RTP/AVP 18 0 101
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv




[7] 2011/08/17 10:55:36:

SIP Tr udp:208.115.60.142:5060:



INVITE sip:1561427xxxx@208.115.60.142;user=phone SIP/2.0
Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-5e0d32504503372356c54f08ea6ef395;rport
From: <sip:5617431521@208.115.60.142>;tag=174310263
To: <sip:1561427xxxx@208.115.60.142;user=phone>
Call-ID: a086938b@pbx
CSeq: 14030 INVITE
Max-Forwards: 70
Contact: <sip:5617431521@208.115.60.136:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: AllstateTelecom-PBX/4.0.1.3499
Proxy-Authorization: Digest realm="208.115.60.142",nonce="4e4bd777000004bf7965b8ab80081eb702aeab3b0d6c916b",response="643e10a3e747539004c22ef2b9175779",username="5617431521",uri="sip:1561427xxxx@208.115.60.142;user=phone",algorithm=MD5
Content-Type: application/sdp
Content-Length: 290

v=0
o=- 1066957623 1066957623 IN IP4 208.115.60.136
s=-
c=IN IP4 208.115.60.136
t=0 0
m=audio 61400 RTP/AVP 18 0 101
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv




[7] 2011/08/17 10:55:37:

SIP Rx tls:64.139.77.137:54696:



PRACK sip:405@208.115.60.136:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-s7rdxl5fra9l;rport
From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg
To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone>;tag=aa112c7956
Call-ID: 3c4b75252475-v0yf8yotze2n
CSeq: 2 PRACK
Max-Forwards: 70
Contact: <sip:405@192.168.100.120:1402;transport=tls;line=5z91mad1>;reg-id=1
RAck: 1 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Proxy-Require: buttons
Content-Length: 0





[7] 2011/08/17 10:55:37:

SIP Tx tls:64.139.77.137:54696:



SIP/2.0 200 Ok
Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-s7rdxl5fra9l;rport=54696;received=64.139.77.137
From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg
To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone>;tag=aa112c7956
Call-ID: 3c4b75252475-v0yf8yotze2n
CSeq: 2 PRACK
Contact: <sip:405@208.115.60.136:5061;transport=tls>
User-Agent: AllstateTelecom-PBX/4.0.1.3499
Content-Length: 0





[7] 2011/08/17 10:55:37:

Discard SRTCP packet from 64.139.77.137:3725 with wrong MAC



[6] 2011/08/17 10:55:37:

Sending RTP for 3c4b75252475-v0yf8yotze2n#aa112c7956 to 64.139.77.137:3724



[7] 2011/08/17 10:55:37:

SIP Rx udp:208.115.60.142:5060:



SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-5e0d32504503372356c54f08ea6ef395;rport=5060
From: <sip:5617431521@208.115.60.142>;tag=174310263
To: <sip:1561427xxxx@208.115.60.142;user=phone>
Call-ID: a086938b@pbx
CSeq: 14030 INVITE
Server: OpenSIPS (1.6.4-2-tls (i386/linux))
Content-Length: 0





[7] 2011/08/17 10:55:37:

SIP Rx udp:208.115.60.142:5060:



SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-5e0d32504503372356c54f08ea6ef395;rport=5060
From: <sip:5617431521@208.115.60.142>;tag=174310263
To: <sip:1561427xxxx@208.115.60.142;user=phone>;tag=7696ca4181be7f5787517527c78a3092.0d2d
Call-ID: a086938b@pbx
CSeq: 14030 INVITE
Proxy-Authenticate: Digest realm="208.115.60.142", nonce="4e4bd779000004c0e6af14460e0b7de18ace196d8577917f", stale=true
Server: OpenSIPS (1.6.4-2-tls (i386/linux))
Content-Length: 0





[7] 2011/08/17 10:55:37:

Call a086938b@pbx#174310263: Clear last INVITE



[7] 2011/08/17 10:55:37:

SIP Tx udp:208.115.60.142:5060:



ACK sip:1561427xxxx@208.115.60.142;user=phone SIP/2.0
Via: SIP/2.0/UDP 208.115.60.136:5060;branch=z9hG4bK-5e0d32504503372356c54f08ea6ef395;rport
From: <sip:5617431521@208.115.60.142>;tag=174310263
To: <sip:1561427xxxx@208.115.60.142;user=phone>;tag=7696ca4181be7f5787517527c78a3092.0d2d
Call-ID: a086938b@pbx
CSeq: 14030 ACK
Max-Forwards: 70
Contact: <sip:5617431521@208.115.60.136:5060;transport=udp>
Content-Length: 0





[5] 2011/08/17 10:55:37:

INVITE Response 407 Proxy Authentication Required: Terminate a086938b@pbx



[7] 2011/08/17 10:55:37:

SIP Tx tls:64.139.77.137:54696:



SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-5sw0ts03k0wc;rport=54696;received=64.139.77.137
From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg
To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone>;tag=aa112c7956
Call-ID: 3c4b75252475-v0yf8yotze2n
CSeq: 1 INVITE
Contact: <sip:405@208.115.60.136:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: AllstateTelecom-PBX/4.0.1.3499
Content-Length: 0





[7] 2011/08/17 10:55:38:

SIP Rx tls:64.139.77.137:54696:



ACK sip:427xxxx@pbx.allstatecomputers.com;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-5sw0ts03k0wc;rport
From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg
To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone>;tag=aa112c7956
Call-ID: 3c4b75252475-v0yf8yotze2n
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:405@192.168.100.120:1402;transport=tls;line=5z91mad1>;reg-id=1
Proxy-Require: buttons
Content-Length: 0





[7] 2011/08/17 10:55:39:

SIP Rx udp:208.115.60.142:5060:



SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 208.115.60.136:5060;received=208.115.60.136;branch=z9hG4bK-5e0d32504503372356c54f08ea6ef395;rport=5060
Record-Route: <sip:sansay527215264rdb19609@64.136.174.30:5060;lr;transport=udp>
Record-Route: <sip:208.115.60.142;lr=on>
To: <sip:1561427xxxx@208.115.60.142;user=phone>;tag=sansay527215264rdb19609
From: <sip:5617431521@208.115.60.142>;tag=174310263
Call-ID: a086938b@pbx
CSeq: 14030 INVITE
Contact: <sip:1561427xxxx@64.136.174.30:5060>
Content-Type: application/sdp
Content-Length: 214

v=0
o=Sansay-VSXi 188 1 IN IP4 64.136.174.30
s=Session Controller
c=IN IP4 74.120.93.70
t=0 0
m=audio 21672 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20




[7] 2011/08/17 10:55:41:

SIP Rx tls:64.139.77.137:54696:



CANCEL sip:427xxxx@pbx.allstatecomputers.com;user=phone SIP/2.0
Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-5sw0ts03k0wc;rport
From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg
To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone>
Call-ID: 3c4b75252475-v0yf8yotze2n
CSeq: 1 CANCEL
Max-Forwards: 70
Reason: SIP;cause=487;text="Request terminated by user"
Proxy-Require: buttons
Content-Length: 0





[7] 2011/08/17 10:55:41:

SIP Tx tls:64.139.77.137:54696:



SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/TLS 192.168.100.120:1402;branch=z9hG4bK-5sw0ts03k0wc;rport=54696;received=64.139.77.137
From: "Brian Artigas" <sip:405@pbx.allstatecomputers.com>;tag=9aa6p7ailg
To: <sip:427xxxx@pbx.allstatecomputers.com;user=phone>
Call-ID: 3c4b75252475-v0yf8yotze2n
CSeq: 1 CANCEL
Content-Length: 0

 

Any input would be greatly appreciated.

 

Thanks,

Brian

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check the obvious stuff.

Has security clobbered the ip addresses ranges

Call me crazy, but our crafted Centos and Windows PBX's have public facing IP addresses,

Are your trunk provider interfaces separate nics? Are they static Routed to the Carrier?

Can you duplicate this with a NON-SNOM, soft - Phone Etc.

While PBX traces are Ok, a passive wireshark trace tells a much more complete story.

 

The Goal in any troubleshooting scenario is to ask questions and take small incremental steps so that the only answer to the question being asked is YES or No.

 

Hopefully you can take a SWAG at the first question so isolate the system into two parts. then take action and when looking at the evidence you can determine is this a problem 100% on the left or right side of where you are looking.

 

Is the problem on the trunk or caller side?

Does the same issue exist with another NIC?

and on, and on, and on,

 

Good Luck. Andy

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