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SIP Issues After Upgrade to 7.0(5)

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BlueRiver31

IS-IT--Management
Dec 11, 2009
66
0
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US
I have a SIP trunk through Voxitas. I upgraded the IP 500 V2 to 7.0(5) from 6.1(12). After the upgrade I found inbound calls take longer to go through and sometimes the call does not go through at all. Some calls take 45 seconds before the phone on the calling party end will start ringing. This issue does not happen on every call. It is very scattered when it happens. Any suggestions would be greatly appreciated.

 
Here is a monitor trace of when a call is made and the issue occurs.




2433531969mS CMCallEvt: 0.55429.0 -1 BaseEP: NEW CMEndpoint f51be0f0 TOTAL NOW=3 CALL_LIST=1
2433531970mS SIP Tx: UDP 10.10.10.11:5060 -> 206.80.67.51:5060
REGISTER sip:stl06a.netlogic.net SIP/2.0
Via: SIP/2.0/UDP 10.10.10.11:1231;rport;branch=z9hG4bK334f087d4a449195a0db5714f9474b12
From: <sip:2175001@stl06a.netlogic.net>;tag=90a4fca931a19195
To: <sip:2175001@stl06a.netlogic.net>
Call-ID: 53f4b41eb66866ea3190e6949b020450@10.10.10.11
CSeq: 1454898009 REGISTER
Contact: "Unknown" <sip:2175001@10.10.10.11:1231;transport=udp>
Expires: 180
Max-Forwards: 70
User-Agent: IP Office 7.0 (5)
Supported: timer
Content-Length: 0

2433532000mS SIP Rx: UDP 206.80.67.51:5060 -> 10.10.10.11:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.11:1231;branch=z9hG4bK334f087d4a449195a0db5714f9474b12;received=10.10.10.11;rport=1231
From: <sip:2175001@stl06a.netlogic.net>;tag=90a4fca931a19195
To: <sip:2175001@stl06a.netlogic.net>
Call-ID: 53f4b41eb66866ea3190e6949b020450@10.10.10.11
CSeq: 1454898009 REGISTER
User-Agent: NetLogic Switch v3.2.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2175001@206.80.67.51>
Content-Length: 0

2433532004mS SIP Rx: UDP 206.80.67.51:5060 -> 10.10.10.11:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.10.10.11:1231;branch=z9hG4bK334f087d4a449195a0db5714f9474b12;received=10.10.10.11;rport=1231
From: <sip:2175001@stl06a.netlogic.net>;tag=90a4fca931a19195
To: <sip:2175001@stl06a.netlogic.net>;tag=as146ace06
Call-ID: 53f4b41eb66866ea3190e6949b020450@10.10.10.11
CSeq: 1454898009 REGISTER
User-Agent: NetLogic Switch v3.2.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Digest algorithm=MD5, realm="asterisk", nonce="2427fb00"
Content-Length: 0

2433532007mS SIP Tx: UDP 10.10.10.11:5060 -> 206.80.67.51:5060
REGISTER sip:stl06a.netlogic.net SIP/2.0
Via: SIP/2.0/UDP 10.10.10.11:1231;rport;branch=z9hG4bK64718e382433a56089c1b0e8157cc526
From: <sip:2175001@stl06a.netlogic.net>;tag=90a4fca931a19195
To: <sip:2175001@stl06a.netlogic.net>
Call-ID: 53f4b41eb66866ea3190e6949b020450@10.10.10.11
CSeq: 1454898010 REGISTER
Contact: "Unknown" <sip:2175001@10.10.10.11:1231;transport=udp>
Expires: 180
Authorization: Digest username="2175001",realm="asterisk",nonce="2427fb00",response="69d22add46c004bddf18273fac87a2f4",uri="sip:stl06a.netlogic.net"
Max-Forwards: 70
User-Agent: IP Office 7.0 (5)
Supported: timer
Content-Length: 0

2433532037mS SIP Rx: UDP 206.80.67.51:5060 -> 10.10.10.11:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.11:1231;branch=z9hG4bK64718e382433a56089c1b0e8157cc526;received=10.10.10.11;rport=1231
From: <sip:2175001@stl06a.netlogic.net>;tag=90a4fca931a19195
To: <sip:2175001@stl06a.netlogic.net>
Call-ID: 53f4b41eb66866ea3190e6949b020450@10.10.10.11
CSeq: 1454898010 REGISTER
User-Agent: NetLogic Switch v3.2.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2175001@206.80.67.51>
Content-Length: 0

2433532041mS SIP Rx: UDP 206.80.67.51:5060 -> 10.10.10.11:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.11:1231;branch=z9hG4bK64718e382433a56089c1b0e8157cc526;received=10.10.10.11;rport=1231
From: <sip:2175001@stl06a.netlogic.net>;tag=90a4fca931a19195
To: <sip:2175001@stl06a.netlogic.net>;tag=as146ace06
Call-ID: 53f4b41eb66866ea3190e6949b020450@10.10.10.11
CSeq: 1454898010 REGISTER
User-Agent: NetLogic Switch v3.2.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 180
Contact: <sip:2175001@10.10.10.11:1231;transport=udp>;expires=180
Date: Tue, 17 May 2011 15:30:02 GMT
Content-Length: 0

2433532044mS CMCallEvt: 0.55429.0 -1 SIPTrunk Endpoint: StateChange: END=X CMCSIdle->CMCSDelete
2433532045mS CMCallEvt: 0.55429.0 -1 BaseEP: DELETE CMEndpoint f51be0f0 TOTAL NOW=2 CALL_LIST=1
2433532195mS RES: Tue 17/5/2011 11:30:02 FreeMem=61849368(1) CMMsg=8 (9) Buff=200 955 999 7463 5 Links=19234
2433532195mS RES2: IP 500 V2 7.0(5) Tasks=44 RTEngine=0 CMRTEngine=0 ExRTEngine=0 Timer=76 Poll=0 Ready=0 CMReady=0 CMQueue=0 VPNNQueue=0 Monitor=1 SSA=1 ASC=1 SYS=MNTD OPT=UMNT SDSPD=2034
2433534727mS SIP Rx: UDP 206.80.67.51:5060 -> 10.10.10.11:5060
CANCEL sip:3177808344@69.54.72.39:1231 SIP/2.0
Via: SIP/2.0/UDP 206.80.67.51:5060;branch=z9hG4bK60acc50c;rport
From: "BOYER NETWORKS " <sip:+18127561746@206.80.67.51>;tag=as0a191fd9
To: <sip:3177808344@69.54.72.39:1231>
Call-ID: 6fa0111705f194b8569f09c9478fba94@206.80.67.51
CSeq: 102 CANCEL
User-Agent: NetLogic Switch v3.2.3
Max-Forwards: 70
Content-Length: 0

2433535733mS SIP Rx: UDP 206.80.67.51:5060 -> 10.10.10.11:5060
CANCEL sip:3177808344@69.54.72.39:1231 SIP/2.0
Via: SIP/2.0/UDP 206.80.67.51:5060;branch=z9hG4bK60acc50c;rport
From: "BOYER NETWORKS " <sip:+18127561746@206.80.67.51>;tag=as0a191fd9
To: <sip:3177808344@69.54.72.39:1231>
Call-ID: 6fa0111705f194b8569f09c9478fba94@206.80.67.51
CSeq: 102 CANCEL
User-Agent: NetLogic Switch v3.2.3
Max-Forwards: 70
Content-Length: 0

2433536734mS SIP Rx: UDP 206.80.67.51:5060 -> 10.10.10.11:5060
CANCEL sip:3177808344@69.54.72.39:1231 SIP/2.0
Via: SIP/2.0/UDP 206.80.67.51:5060;branch=z9hG4bK60acc50c;rport
From: "BOYER NETWORKS " <sip:+18127561746@206.80.67.51>;tag=as0a191fd9
To: <sip:3177808344@69.54.72.39:1231>
Call-ID: 6fa0111705f194b8569f09c9478fba94@206.80.67.51
CSeq: 102 CANCEL
User-Agent: NetLogic Switch v3.2.3
Max-Forwards: 70
Content-Length: 0
 
option 1:
Create a extra URI, instead of "Use Credentials User Name" fill in the phone number in the Local URI/Contact/Display Name.

Give it a new outgoing group id, and change the ARS to that group id.

Option2:
On the users SIP tab, change the SIP name/SIP Display name (alias)/Contact into the number you would like to send.

Extra;
For incomming you can add a * on all 3 pull down menus instead of "Use Credentials User Name" fill in the phone number in the Local URI/Contact/Display Name. This way you don't need to put in every SIP number to make it ring on incomming calls.

Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool )
______________________________________
 
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