[SR-Users] BYE message not relayed to caller
oneten
go at onetennetworks.com
Tue Dec 4 23:25:58 CET 2012
Vikram Ragukumar wrote:
>
> Daniel,
>
> Thank you for your response.
>
> We have verified that it is indeed a bug with VoipSwitch. We uninstalled
> the older version (v2.0.0965) of VoipSwitch that we were running and
> replaced it with a newer version (v2.985), and the problem went away.
>
> Once again, thank you very much for your assistance.
>
> Regards,
> Vikram.
>
>>
>>
>> On 02/24/2010 08:43 PM, Vikram Ragukumar wrote:
>>> Hello,
>>>
>>> In the Call flow diagram Phone B is to be read as VoipSwitch.
>>>
>>
>> ignore previous email, I read that first and replied immediately ...
>>
>> From the sip trace, the INVITE going to B has good record-route and
>> contact header. Therefore looks to be a bug in voipswitch.
>>
>> Daniel
>>
>>> Regards,
>>> Vikram.
>>>
>>>> Daniel,
>>>>
>>>> I have tried to summarize the SIP message flow below. I am also
>>>> including
>>>> the entire SIP trace at the end of this message.
>>>>
>>>> Cell Phone Kamailio Phone B
>>>> | | |
>>>> |INVITE | |
>>>> |------------->| |
>>>> |100 Trying | |
>>>> |<-------------| |
>>>> | |INVITE |
>>>> | |------------->|
>>>> | |100 trying |
>>>> | |<-------------|
>>>> | |183SessionProg|
>>>> | |<-------------|
>>>> |183SessionProg| |
>>>> |<-------------| |
>>>> | | 200 OK |
>>>> | 200 OK |<-------------|
>>>> |<-------------| |
>>>> | ACK | |
>>>> |------------->| |
>>>> | | ACK |
>>>> | |------------->|
>>>> |200 OK | |
>>>> |<-------------| |
>>>> | | BYE |
>>>> | |<-------------|<-
>>>> BYE,RURI=account at VoipSwitch
>>>> | | BYE |
>>>> | |------------->|
>>>> | | BYE |
>>>> | |------------->|
>>>>
>>>>
>>>> What might be causing VoipSwitch to send a BYE with
>>>> RURI=account at VoipSwitch?
>>>> As a result the BYE message never gets forwarded to the cellphone, and
>>>> the
>>>> proxy repeatedly sends BYE messages back to VoipSwitch.
>>>>
>>>> Thanks in advance for your help.
>>>> Regards,
>>>> Vikram.
>>>>
>>>> PS : Below is the SIP trace for the above call flow.
>>>>
>>>> ----------------------------------------------------------------------------
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 16 5.676114 Cell_phone_gw Proxy SIP/SDP
>>>> Request:
>>>> INVITE sip:1234 at VoipSwitch:5060, with session description
>>>>
>>>> Frame 16 (1264 bytes on wire, 1264 bytes captured)
>>>> Ethernet II, Src: 00:26:f2:c8:44:11 (00:26:f2:c8:44:11), Dst:
>>>> Supermic_bd:b9:bc (00:30:48:bd:b9:bc)
>>>> Internet Protocol, Src: Cell_phone_gw (Cell_phone_gw), Dst: Proxy
>>>> (Proxy)
>>>> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
>>>> Session Initiation Protocol
>>>> Request-Line: INVITE sip:1234 at VoipSwitch:5060 SIP/2.0
>>>> Message Header
>>>> Via: SIP/2.0/UDP
>>>> 192.168.1.101:5060;rport;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW
>>>> Max-Forwards: 70
>>>> From: "91131"
>>>> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>>>> To: sip:1234 at VoipSwitch
>>>> Contact: "91131"<sip:91131 at 192.168.1.101:5060>
>>>> Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>>>> CSeq: 24680 INVITE
>>>> Route:<sip:Proxy:5060;lr>
>>>> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE,
>>>> NOTIFY,
>>>> REFER, MESSAGE, OPTIONS
>>>> Supported: replaces, 100rel, timer, norefersub
>>>> Session-Expires: 1800
>>>> Min-SE: 90
>>>> Proxy-Authorization: Digest username="91131",
>>>> realm="VoipSwitch",
>>>> nonce="126686109922231105302513908108",
>>>> uri="sip:1234 at VoipSwitch:5060",
>>>> response="55122bcb903503303164237e62481f52"
>>>> Content-Type: application/sdp
>>>> Content-Length: 379
>>>> Message Body
>>>> Session Description Protocol
>>>> Session Description Protocol Version (v): 0
>>>> Owner/Creator, Session Id (o): - 3475932668 3475932668 IN
>>>> IP4
>>>> 192.168.1.101
>>>> Session Name (s): pjmedia
>>>> Connection Information (c): IN IP4 192.168.1.101
>>>> Time Description, active time (t): 0 0
>>>> Session Attribute (a): X-nat:0
>>>> Media Description, name and address (m): audio 4000
>>>> RTP/AVP
>>>> 114 18 113 0 8 101
>>>> Media Attribute (a): rtcp:4001 IN IP4 192.168.1.101
>>>> Media Attribute (a): rtpmap:114 AMR/8000
>>>> Media Attribute (a): rtpmap:18 G729/8000
>>>> Media Attribute (a): rtpmap:113 iLBC/8000
>>>> Media Attribute (a): fmtp:113 mode=30
>>>> Media Attribute (a): rtpmap:0 PCMU/8000
>>>> Media Attribute (a): rtpmap:8 PCMA/8000
>>>> Media Attribute (a): sendrecv
>>>> Media Attribute (a): rtpmap:101 telephone-event/8000
>>>> Media Attribute (a): fmtp:101 0-15
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 17 5.744897 Proxy Cell_phone_gw SIP Status:
>>>> 100
>>>> Giving a try
>>>>
>>>> Frame 17 (429 bytes on wire, 429 bytes captured)
>>>> Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst:
>>>> 00:26:f2:c8:44:11 (00:26:f2:c8:44:11)
>>>> Internet Protocol, Src: Proxy (Proxy), Dst: Cell_phone_gw
>>>> (Cell_phone_gw)
>>>> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
>>>> Session Initiation Protocol
>>>> Status-Line: SIP/2.0 100 Giving a try
>>>> Message Header
>>>> Via: SIP/2.0/UDP
>>>> 192.168.1.101:5060;rport=5060;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW;received=Cell_phone_gw
>>>> From: "91131"
>>>> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>>>> To: sip:1234 at VoipSwitch
>>>> Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>>>> CSeq: 24680 INVITE
>>>> Server: Kamailio (1.5.3-notls (i386/linux))
>>>> Content-Length: 0
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 18 5.747037 Proxy VoipSwitch SIP/SDP Request:
>>>> INVITE sip:1234 at VoipSwitch:5060, with session description
>>>>
>>>> Frame 18 (1434 bytes on wire, 1434 bytes captured)
>>>> Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst:
>>>> Unispher_40:b5:39 (00:90:1a:40:b5:39)
>>>> Internet Protocol, Src: Proxy (Proxy), Dst: VoipSwitch (VoipSwitch)
>>>> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: sip (5060)
>>>> Session Initiation Protocol
>>>> Request-Line: INVITE sip:1234 at VoipSwitch:5060 SIP/2.0
>>>> Message Header
>>>> Record-Route:<sip:Proxy:5060;lr=on;nat=yes>
>>>> Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKd5eb.409fb37.0
>>>> Via: SIP/2.0/UDP
>>>> 192.168.1.101:5060;received=Cell_phone_gw;rport=5060;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW
>>>> Max-Forwards: 69
>>>> From: "91131"
>>>> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>>>> To: sip:1234 at VoipSwitch
>>>> Contact: "91131"<sip:91131 at Cell_phone_gw:5060>
>>>> Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>>>> CSeq: 24680 INVITE
>>>> Route:<sip:Proxy:5060;lr>
>>>> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE,
>>>> NOTIFY,
>>>> REFER, MESSAGE, OPTIONS
>>>> Supported: replaces, 100rel, timer, norefersub
>>>> Session-Expires: 1800
>>>> Min-SE: 90
>>>> Proxy-Authorization: Digest username="91131",
>>>> realm="VoipSwitch",
>>>> nonce="126686109922231105302513908108",
>>>> uri="sip:1234 at VoipSwitch:5060",
>>>> response="55122bcb903503303164237e62481f52"
>>>> Content-Type: application/sdp
>>>> Content-Length: 379
>>>> P-hint: outbound
>>>> Message Body
>>>> Session Description Protocol
>>>> Session Description Protocol Version (v): 0
>>>> Owner/Creator, Session Id (o): - 3475932668 3475932668 IN
>>>> IP4
>>>> 192.168.1.101
>>>> Session Name (s): pjmedia
>>>> Connection Information (c): IN IP4 Proxy
>>>> Time Description, active time (t): 0 0
>>>> Session Attribute (a): X-nat:0
>>>> Media Description, name and address (m): audio 35752
>>>> RTP/AVP
>>>> 114 18 113 0 8 101
>>>> Media Attribute (a): rtcp:35753
>>>> Media Attribute (a): rtpmap:114 AMR/8000
>>>> Media Attribute (a): rtpmap:18 G729/8000
>>>> Media Attribute (a): rtpmap:113 iLBC/8000
>>>> Media Attribute (a): fmtp:113 mode=30
>>>> Media Attribute (a): rtpmap:0 PCMU/8000
>>>> Media Attribute (a): rtpmap:8 PCMA/8000
>>>> Media Attribute (a): sendrecv
>>>> Media Attribute (a): rtpmap:101 telephone-event/8000
>>>> Media Attribute (a): fmtp:101 0-15
>>>> Media Attribute (a): nortpproxy:yes
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 19 5.934950 VoipSwitch Proxy SIP Status:
>>>> 100
>>>> Trying
>>>>
>>>> Frame 19 (579 bytes on wire, 579 bytes captured)
>>>> Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst:
>>>> Supermic_bd:b9:bc (00:30:48:bd:b9:bc)
>>>> Internet Protocol, Src: VoipSwitch (VoipSwitch), Dst: Proxy (Proxy)
>>>> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5060 (5060)
>>>> Session Initiation Protocol
>>>> Status-Line: SIP/2.0 100 Trying
>>>> Message Header
>>>> CSeq: 24680 INVITE
>>>> Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKd5eb.409fb37.0
>>>> Via: SIP/2.0/UDP
>>>> 192.168.1.101:5060;received=Cell_phone_gw;rport=5060;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW
>>>> From: "91131"
>>>> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>>>> Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>>>> To: sip:1234 at VoipSwitch;tag=22025110233933268788916305
>>>> Contact:<sip:VoipSwitch:5060;transport=udp>
>>>> Content-Length: 0
>>>> Record-Route:<sip:Proxy:5060;lr=on;nat=yes>
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 20 6.707560 VoipSwitch Proxy SIP/SDP Status:
>>>> 183
>>>> Session Progress, with session description
>>>>
>>>> Frame 20 (868 bytes on wire, 868 bytes captured)
>>>> Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst:
>>>> Supermic_bd:b9:bc (00:30:48:bd:b9:bc)
>>>> Internet Protocol, Src: VoipSwitch (VoipSwitch), Dst: Proxy (Proxy)
>>>> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5060 (5060)
>>>> Session Initiation Protocol
>>>> Status-Line: SIP/2.0 183 Session Progress
>>>> Message Header
>>>> CSeq: 24680 INVITE
>>>> Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKd5eb.409fb37.0
>>>> Via: SIP/2.0/UDP
>>>> 192.168.1.101:5060;received=Cell_phone_gw;rport=5060;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW
>>>> From: "91131"
>>>> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>>>> Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>>>> To: sip:1234 at VoipSwitch;tag=22025110233933268788916305
>>>> Contact:<sip:VoipSwitch:5060;transport=udp>
>>>> Content-Type: application/sdp
>>>> Content-Length: 246
>>>> Record-Route:<sip:Proxy:5060;lr=on;nat=yes>
>>>> Message Body
>>>> Session Description Protocol
>>>> Session Description Protocol Version (v): 0
>>>> Owner/Creator, Session Id (o): VoipSwitch 7304 7304 IN IP4
>>>> VoipSwitch
>>>> Session Name (s): VoipSIP
>>>> Session Information (i): Audio Session
>>>> Connection Information (c): IN IP4 VoipSwitch
>>>> Time Description, active time (t): 0 0
>>>> Media Description, name and address (m): audio 6304
>>>> RTP/AVP 18
>>>> 101
>>>> Media Attribute (a): rtpmap:18 G729/8000/1
>>>> Media Attribute (a): fmtp:18 annexb=no
>>>> Media Attribute (a): rtpmap:101 telephone-event/8000
>>>> Media Attribute (a): fmtp:101 0-15
>>>> Media Attribute (a): sendrecv
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 21 6.734267 Proxy Cell_phone_gw SIP/SDP Status:
>>>> 183
>>>> Session Progress, with session description
>>>>
>>>> Frame 21 (822 bytes on wire, 822 bytes captured)
>>>> Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst:
>>>> 00:26:f2:c8:44:11 (00:26:f2:c8:44:11)
>>>> Internet Protocol, Src: Proxy (Proxy), Dst: Cell_phone_gw
>>>> (Cell_phone_gw)
>>>> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
>>>> Session Initiation Protocol
>>>> Status-Line: SIP/2.0 183 Session Progress
>>>> Message Header
>>>> CSeq: 24680 INVITE
>>>> Via: SIP/2.0/UDP
>>>> 192.168.1.101:5060;received=Cell_phone_gw;rport=5060;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW
>>>> From: "91131"
>>>> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>>>> Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>>>> To: sip:1234 at VoipSwitch;tag=22025110233933268788916305
>>>> Contact:<sip:VoipSwitch:5060;transport=udp>
>>>> Content-Type: application/sdp
>>>> Content-Length: 267
>>>> Record-Route:<sip:Proxy:5060;lr=on;nat=yes>
>>>> Message Body
>>>> Session Description Protocol
>>>> Session Description Protocol Version (v): 0
>>>> Owner/Creator, Session Id (o): VoipSwitch 7304 7304 IN IP4
>>>> VoipSwitch
>>>> Session Name (s): VoipSIP
>>>> Session Information (i): Audio Session
>>>> Connection Information (c): IN IP4 Proxy
>>>> Time Description, active time (t): 0 0
>>>> Media Description, name and address (m): audio 35570
>>>> RTP/AVP
>>>> 18 101
>>>> Media Attribute (a): rtpmap:18 G729/8000/1
>>>> Media Attribute (a): fmtp:18 annexb=no
>>>> Media Attribute (a): rtpmap:101 telephone-event/8000
>>>> Media Attribute (a): fmtp:101 0-15
>>>> Media Attribute (a): sendrecv
>>>> Media Attribute (a): nortpproxy:yes
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 22 15.889935 Cell_phone_gw Proxy UDP Source
>>>> port: 5060 Destination port: 5060
>>>>
>>>> Frame 22 (60 bytes on wire, 60 bytes captured)
>>>> Ethernet II, Src: 00:26:f2:c8:44:11 (00:26:f2:c8:44:11), Dst:
>>>> Supermic_bd:b9:bc (00:30:48:bd:b9:bc)
>>>> Internet Protocol, Src: Cell_phone_gw (Cell_phone_gw), Dst: Proxy
>>>> (Proxy)
>>>> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
>>>> Data (2 bytes)
>>>>
>>>> 0000 0d 0a ..
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 23 19.801513 VoipSwitch Proxy SIP/SDP Status:
>>>> 200
>>>> OK, with session description
>>>>
>>>> Frame 23 (854 bytes on wire, 854 bytes captured)
>>>> Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst:
>>>> Supermic_bd:b9:bc (00:30:48:bd:b9:bc)
>>>> Internet Protocol, Src: VoipSwitch (VoipSwitch), Dst: Proxy (Proxy)
>>>> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5060 (5060)
>>>> Session Initiation Protocol
>>>> Status-Line: SIP/2.0 200 OK
>>>> Message Header
>>>> CSeq: 24680 INVITE
>>>> Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKd5eb.409fb37.0
>>>> Via: SIP/2.0/UDP
>>>> 192.168.1.101:5060;received=Cell_phone_gw;rport=5060;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW
>>>> From: "91131"
>>>> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>>>> Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>>>> To: sip:1234 at VoipSwitch;tag=22025110233933268788916305
>>>> Contact:<sip:VoipSwitch:5060;transport=udp>
>>>> Content-Type: application/sdp
>>>> Content-Length: 246
>>>> Record-Route:<sip:Proxy:5060;lr=on;nat=yes>
>>>> Message Body
>>>> Session Description Protocol
>>>> Session Description Protocol Version (v): 0
>>>> Owner/Creator, Session Id (o): VoipSwitch 7304 7304 IN IP4
>>>> VoipSwitch
>>>> Session Name (s): VoipSIP
>>>> Session Information (i): Audio Session
>>>> Connection Information (c): IN IP4 VoipSwitch
>>>> Time Description, active time (t): 0 0
>>>> Media Description, name and address (m): audio 6304
>>>> RTP/AVP 18
>>>> 101
>>>> Media Attribute (a): rtpmap:18 G729/8000/1
>>>> Media Attribute (a): fmtp:18 annexb=no
>>>> Media Attribute (a): rtpmap:101 telephone-event/8000
>>>> Media Attribute (a): fmtp:101 0-15
>>>> Media Attribute (a): sendrecv
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 24 19.851387 Proxy Cell_phone_gw SIP/SDP Status:
>>>> 200
>>>> OK, with session description
>>>>
>>>> Frame 24 (808 bytes on wire, 808 bytes captured)
>>>> Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst:
>>>> 00:26:f2:c8:44:11 (00:26:f2:c8:44:11)
>>>> Internet Protocol, Src: Proxy (Proxy), Dst: Cell_phone_gw
>>>> (Cell_phone_gw)
>>>> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
>>>> Session Initiation Protocol
>>>> Status-Line: SIP/2.0 200 OK
>>>> Message Header
>>>> CSeq: 24680 INVITE
>>>> Via: SIP/2.0/UDP
>>>> 192.168.1.101:5060;received=Cell_phone_gw;rport=5060;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW
>>>> From: "91131"
>>>> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>>>> Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>>>> To: sip:1234 at VoipSwitch;tag=22025110233933268788916305
>>>> Contact:<sip:VoipSwitch:5060;transport=udp>
>>>> Content-Type: application/sdp
>>>> Content-Length: 267
>>>> Record-Route:<sip:Proxy:5060;lr=on;nat=yes>
>>>> Message Body
>>>> Session Description Protocol
>>>> Session Description Protocol Version (v): 0
>>>> Owner/Creator, Session Id (o): VoipSwitch 7304 7304 IN IP4
>>>> VoipSwitch
>>>> Session Name (s): VoipSIP
>>>> Session Information (i): Audio Session
>>>> Connection Information (c): IN IP4 Proxy
>>>> Time Description, active time (t): 0 0
>>>> Media Description, name and address (m): audio 35570
>>>> RTP/AVP
>>>> 18 101
>>>> Media Attribute (a): rtpmap:18 G729/8000/1
>>>> Media Attribute (a): fmtp:18 annexb=no
>>>> Media Attribute (a): rtpmap:101 telephone-event/8000
>>>> Media Attribute (a): fmtp:101 0-15
>>>> Media Attribute (a): sendrecv
>>>> Media Attribute (a): nortpproxy:yes
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 25 19.860918 Cell_phone_gw Proxy SIP
>>>> Request:
>>>> ACK sip:VoipSwitch:5060;transport=udp
>>>>
>>>> Frame 25 (470 bytes on wire, 470 bytes captured)
>>>> Ethernet II, Src: 00:26:f2:c8:44:11 (00:26:f2:c8:44:11), Dst:
>>>> Supermic_bd:b9:bc (00:30:48:bd:b9:bc)
>>>> Internet Protocol, Src: Cell_phone_gw (Cell_phone_gw), Dst: Proxy
>>>> (Proxy)
>>>> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
>>>> Session Initiation Protocol
>>>> Request-Line: ACK sip:VoipSwitch:5060;transport=udp SIP/2.0
>>>> Message Header
>>>> Via: SIP/2.0/UDP
>>>> 192.168.1.101:5060;rport;branch=z9hG4bKPj3R8VFolrcYnGiHZ65Foh4rL9rghVDXUW
>>>> Max-Forwards: 70
>>>> From: "91131"
>>>> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>>>> To: sip:1234 at VoipSwitch;tag=22025110233933268788916305
>>>> Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>>>> CSeq: 24680 ACK
>>>> Route:<sip:Proxy:5060;lr;nat=yes>
>>>> Content-Length: 0
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 26 19.901346 Proxy VoipSwitch SIP Request:
>>>> ACK
>>>> sip:VoipSwitch:5060;transport=udp
>>>>
>>>> Frame 26 (521 bytes on wire, 521 bytes captured)
>>>> Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst:
>>>> Unispher_40:b5:39 (00:90:1a:40:b5:39)
>>>> Internet Protocol, Src: Proxy (Proxy), Dst: VoipSwitch (VoipSwitch)
>>>> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: sip (5060)
>>>> Session Initiation Protocol
>>>> Request-Line: ACK sip:VoipSwitch:5060;transport=udp SIP/2.0
>>>> Message Header
>>>> Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKd5eb.409fb37.2
>>>> Via: SIP/2.0/UDP
>>>> 192.168.1.101:5060;received=Cell_phone_gw;rport=5060;branch=z9hG4bKPj3R8VFolrcYnGiHZ65Foh4rL9rghVDXUW
>>>> Max-Forwards: 69
>>>> From: "91131"
>>>> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>>>> To: sip:1234 at VoipSwitch;tag=22025110233933268788916305
>>>> Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>>>> CSeq: 24680 ACK
>>>> Content-Length: 0
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 27 27.987188 VoipSwitch Proxy SIP Request:
>>>> BYE
>>>> sip:91131 at VoipSwitch
>>>>
>>>> Frame 27 (420 bytes on wire, 420 bytes captured)
>>>> Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst:
>>>> Supermic_bd:b9:bc (00:30:48:bd:b9:bc)
>>>> Internet Protocol, Src: VoipSwitch (VoipSwitch), Dst: Proxy (Proxy)
>>>> User Datagram Protocol, Src Port: sip (5060), Dst Port: 5060 (5060)
>>>> Session Initiation Protocol
>>>> Request-Line: BYE sip:91131 at VoipSwitch SIP/2.0
>>>> Message Header
>>>> Route:<sip:Proxy:5060;lr=on;nat=yes>
>>>> CSeq: 1 BYE
>>>> Via: SIP/2.0/UDP
>>>> VoipSwitch:5060;branch=z9hG4bk220252102301223326901297
>>>> From: sip:1234 at VoipSwitch;tag=22025110233933268788916305
>>>> Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>>>> To: "91131"
>>>> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>>>> Content-Length: 0
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 28 28.211030 Proxy VoipSwitch SIP Request:
>>>> BYE
>>>> sip:91131 at VoipSwitch
>>>>
>>>> Frame 28 (490 bytes on wire, 490 bytes captured)
>>>> Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst:
>>>> Unispher_40:b5:39 (00:90:1a:40:b5:39)
>>>> Internet Protocol, Src: Proxy (Proxy), Dst: VoipSwitch (VoipSwitch)
>>>> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: sip (5060)
>>>> Session Initiation Protocol
>>>> Request-Line: BYE sip:91131 at VoipSwitch SIP/2.0
>>>> Message Header
>>>> Max-Forwards: 10
>>>> CSeq: 1 BYE
>>>> Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKa0bd.f0181ba.0
>>>> Via: SIP/2.0/UDP
>>>> VoipSwitch:5060;rport=5060;received=VoipSwitch;branch=z9hG4bk220252102301223326901297
>>>> From: sip:1234 at VoipSwitch;tag=22025110233933268788916305
>>>> Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>>>> To: "91131"
>>>> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>>>> Content-Length: 0
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 29 28.698172 Proxy VoipSwitch SIP Request:
>>>> BYE
>>>> sip:91131 at VoipSwitch
>>>>
>>>> Frame 29 (490 bytes on wire, 490 bytes captured)
>>>> Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst:
>>>> Unispher_40:b5:39 (00:90:1a:40:b5:39)
>>>> Internet Protocol, Src: Proxy (Proxy), Dst: VoipSwitch (VoipSwitch)
>>>> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: sip (5060)
>>>> Session Initiation Protocol
>>>> Request-Line: BYE sip:91131 at VoipSwitch SIP/2.0
>>>> Message Header
>>>> Max-Forwards: 10
>>>> CSeq: 1 BYE
>>>> Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKa0bd.f0181ba.0
>>>> Via: SIP/2.0/UDP
>>>> VoipSwitch:5060;rport=5060;received=VoipSwitch;branch=z9hG4bk220252102301223326901297
>>>> From: sip:1234 at VoipSwitch;tag=22025110233933268788916305
>>>> Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>>>> To: "91131"
>>>> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>>>> Content-Length: 0
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 30 29.698214 Proxy VoipSwitch SIP Request:
>>>> BYE
>>>> sip:91131 at VoipSwitch
>>>>
>>>> Frame 30 (490 bytes on wire, 490 bytes captured)
>>>> Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst:
>>>> Unispher_40:b5:39 (00:90:1a:40:b5:39)
>>>> Internet Protocol, Src: Proxy (Proxy), Dst: VoipSwitch (VoipSwitch)
>>>> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: sip (5060)
>>>> Session Initiation Protocol
>>>> Request-Line: BYE sip:91131 at VoipSwitch SIP/2.0
>>>> Message Header
>>>> Max-Forwards: 10
>>>> CSeq: 1 BYE
>>>> Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKa0bd.f0181ba.0
>>>> Via: SIP/2.0/UDP
>>>> VoipSwitch:5060;rport=5060;received=VoipSwitch;branch=z9hG4bk220252102301223326901297
>>>> From: sip:1234 at VoipSwitch;tag=22025110233933268788916305
>>>> Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>>>> To: "91131"
>>>> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>>>> Content-Length: 0
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 31 30.941201 Cell_phone_gw Proxy UDP Source
>>>> port: 5060 Destination port: 5060
>>>>
>>>> Frame 31 (60 bytes on wire, 60 bytes captured)
>>>> Ethernet II, Src: 00:26:f2:c8:44:11 (00:26:f2:c8:44:11), Dst:
>>>> Supermic_bd:b9:bc (00:30:48:bd:b9:bc)
>>>> Internet Protocol, Src: Cell_phone_gw (Cell_phone_gw), Dst: Proxy
>>>> (Proxy)
>>>> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
>>>> Data (2 bytes)
>>>>
>>>> 0000 0d 0a ..
>>>>
>>>> No. Time Source Destination
>>>> Protocol
>>>> Info
>>>> 32 31.699278 Proxy VoipSwitch SIP Request:
>>>> BYE
>>>> sip:91131 at VoipSwitch
>>>>
>>>> Frame 32 (490 bytes on wire, 490 bytes captured)
>>>> Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst:
>>>> Unispher_40:b5:39 (00:90:1a:40:b5:39)
>>>> Internet Protocol, Src: Proxy (Proxy), Dst: VoipSwitch (VoipSwitch)
>>>> User Datagram Protocol, Src Port: 5060 (5060), Dst Port: sip (5060)
>>>> Session Initiation Protocol
>>>> Request-Line: BYE sip:91131 at VoipSwitch SIP/2.0
>>>> Message Header
>>>> Max-Forwards: 10
>>>> CSeq: 1 BYE
>>>> Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKa0bd.f0181ba.0
>>>> Via: SIP/2.0/UDP
>>>> VoipSwitch:5060;rport=5060;received=VoipSwitch;branch=z9hG4bk220252102301223326901297
>>>> From: sip:1234 at VoipSwitch;tag=22025110233933268788916305
>>>> Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>>>> To: "91131"
>>>> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>>>> Content-Length: 0
>>>> ----------------------------------------------------------------------------
>>>>
>>>>
>>>>>> Hello,
>>>>>> can you post the entire call flow, from initial invite to to the bye.
>>>>>>
>>>> There is some mistake done somewhere in the routing elements. The sip
>>>> trace will help to identify where.
>>>>
>>>>>> Cheers,
>>>>>> Daniel
>>>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Kamailio (OpenSER) - Users mailing list
>>>> Users at lists.kamailio.org
>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>
>>>
>>
>> --
>> Daniel-Constantin Mierla
>> Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
>> * http://www.asipto.com/index.php/sip-router-masterclass/
>>
>>
>
>
> --
> Thanks and Regards,
> Vikram Ragukumar.
>
> _______________________________________________
> Kamailio (OpenSER) - Users mailing list
> Users at lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
>
Did you ever find a work around on this issue? We are getting 183 session
progress messages when going from voip to pstn. Did using v 2.985 fix this?
--
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