Re: [SR-Users] BYE message not relayed to caller

4 Dec 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@VoipSwitch
            |              |     BYE      |
            |              |------------->|
            |              |     BYE      |
            |              |------------->|

 What might be causing VoipSwitch to send a BYE with
 RURI=account@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@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@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@VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
          To: sip:1234@VoipSwitch
          Contact: "91131"<sip:91131@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@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@VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
          To: sip:1234@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@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@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@VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
          To: sip:1234@VoipSwitch
          Contact: "91131"<sip:91131@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@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@VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
          To: sip:1234@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@VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
          To: sip:1234@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@VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
          To: sip:1234@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@VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
          To: sip:1234@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@VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
          To: sip:1234@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@VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
          To: sip:1234@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@VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
          To: sip:1234@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@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@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@VoipSwitch;tag=22025110233933268788916305
          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
          To: "91131"
 <sip:91131@VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
          Content-Length: 0

 No.     Time        Source                Destination
 Protocol
 Info
       28 28.211030   Proxy        VoipSwitch          SIP      Request:
 BYE
 sip:91131@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@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@VoipSwitch;tag=22025110233933268788916305
          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
          To: "91131"
 <sip:91131@VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
          Content-Length: 0

 No.     Time        Source                Destination
 Protocol
 Info
       29 28.698172   Proxy        VoipSwitch          SIP      Request:
 BYE
 sip:91131@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@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@VoipSwitch;tag=22025110233933268788916305
          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
          To: "91131"
 <sip:91131@VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
          Content-Length: 0

 No.     Time        Source                Destination
 Protocol
 Info
       30 29.698214   Proxy        VoipSwitch          SIP      Request:
 BYE
 sip:91131@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@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@VoipSwitch;tag=22025110233933268788916305
          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
          To: "91131"
 <sip:91131@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@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@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@VoipSwitch;tag=22025110233933268788916305
          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
          To: "91131"
 <sip:91131@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(a)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(a)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?

-- 
View this message in context:
http://old.nabble.com/BYE-message-not-relayed-to-caller-tp27632179p34759588…
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Re: [SR-Users] BYE message not relayed to caller