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Call held after answer

Highlighted
Occasional Advisor

Call held after answer

 

Hello,

 

Any insight into this would be very helpful!  I have exhausted my search of Google and beyond… I have no idea how to solve this issue :/

 

I have a great working setup with VVX 300s 400s 500s and a IP 6000, however there is a strange issue going on with an IP 7000 I have.

 

When calling to or from IP 7000

  1. It rings fine with audio on both ends
  2. Press answer on either side
  3. Call is immediately “Held” on IP 7000
  4. Green lights flash on IP 7000 as if the call was placed on hold by the other phone, but call is not on hold by other phone
  5. No audio (probably because on hold)
  6. End call normally

 

NAT settings are correct

 

  • Formatted IP 7000 and reset all config
  • Reinstalled firmware
  • Disconnected from provisioning server
  • Created  manual config from scratch on phone web interface
  • Removed all encryption

 

IP 7000 logs during call:

Start ringing:

003219.281|net  |2|03|NWIF: nw_setlocalhold() - setting local hold (0) 2.

003219.341|net  |2|03|NWIF: nw_setlocalhold() - setting local hold (0) 2.

003219.500|net  |4|03|rtosNetwork: netwTask() - Can't find associated CCB.

Ring:

003220.080|net  |4|03|rtosNetwork net02: netwSend() - sendto() call failed. fd 1123346772 errno=130

Answer:

003220.861|net  |2|03|NWIF: nw_setlocalhold() - setting local hold (0) 2.

003221.600|net  |4|03|rtosNetwork: netwTask() - Can't find associated CCB.

003223.600|net  |4|03|rtosNetwork: netwTask() - Can't find associated CCB.

Then press hang-up:

003227.600|net  |4|03|rtosNetwork: netwTask() - Can't find associated CCB.

003227.985|net  |2|03|NWIF: nw_setlocalhold() - setting local hold (1) 2.

003242.982|sip  |5|03|Can not decode the packet

 

  • Setting local hold (0) should be fine and it works on all other phones I have.
  • Setting local hold (1) after pressing hang-up is weird but is happening after hang-up anyway.
  • I wonder why that send error is in the logs… it looks like an operating system error…. Maybe a driver error?

Placing a call in the opposite direction i get "GetCallOrder Could not find the call" in the IP 7000 log.

 

All configuration is identical to the working VVX and IP 6000 except the line registration.

 

EDIT: UC version 4.0.10.0568 and 4.0.1.13681 were tested and have the same result.

EDIT: rtosNetwork net02: netwSend() - sendto() call failed happens on every ring before pickup

EDIT: Using FreeSwitch 1.4

 

Thanks in advance for the help!!

Adrian

Message 1 of 6
5 REPLIES 5
Highlighted
Polycom Employee & Community Manager

Re: Call held after answer

Hello Adrian,

welcome to the Polycom Community.

We need some traces and SIP logging at 0 in order to check for anything.

 

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 2 of 6
Highlighted
Occasional Advisor

Re: Call held after answer

Hello,

Thank you for the quick reply.  I have changed the hosts, extensions and IPs in the log file to host.host.com, 1111 and 111.111.111.111.   Everything about the host and the IP looked perfectly fine.

 

I did find a strange error that I highlighted in bold below.  It looks like a codec error is causing a held state.

 

The current installed version is 4.0.1.13681

 

Log snippet:

 

000102.847|sip  |3|03|UA Client INVITE INVITE trans state 'proceeding'->'terminated' by 200 resp 0 timeout(0x41825384)
000102.848|sip  |1|03|CreateFailOverProxyList : Reg to Domain 'host.host.com' nPort 5060
000102.848|sip  |1|03|CreateFailOverProxyList : For ACK Request nPort 5060
000102.848|sip  |1|03|doDnsListLookup(tcp): doDnsSrvLookupForARecordList for 'host.host.com' port 5060 returned 1 results
000102.848|sip  |1|03|doDnsListLookup(tcp): result 0 '111.111.111.111' port 5060 isInBound 0
000102.848|sip  |1|03|CreateFailOverProxyList : 'TCP Only' for 'sip.successos.com' port 5060 IP 0 is 111.111.111.111' on tcp port 5060
000102.848|sip  |2|03|CreateFailOverProxyList : Exit 'TCP Only' lookup with 1 IP Addresses
000102.848|sip  |2|03|CreateFailOverProxyList : IP 1 is 111.111.111.111' on tcp port 5060
000102.848|sip  |1|03|CTcp::Send(TCP) address 111.111.111.111 port 5060 can Connect 1
000102.849|sip  |2|03|adjustRetransWhenTimerCreated UA Client INVITE ACK state 'terminated' timeout=65 (0x41825384)
000102.849|sip  |3|03|GetRemotePartyAddress from 'Remote-Party-ID'
000102.849|sip  |3|03|CStkCall::OnEvNewDest Unchanged display '1111' user ''
000102.849|sip  |2|03|CStateInviteClient::OnEvResponse Normal case
000102.849|sip  |3|03|CStkCall::RemoteSdpAnswer(1) -> ReportCodec( 1)
000102.849|sip  |2|03|CStkCall::ReportCodec: held set true due to m_MediaArray[i].m_inetAddr == 0
000102.849|sip  |3|03|CStkCall::ReportCodec: call state 'RingBack' exit with held 1 (0x418767e4)
000102.850|sip  |1|03|Dialog 'idc86f265e' State 'Early'->'Confirmed'
000102.850|sip  |3|03|CStkCall::NewCallState 'RingBack'->'Held' (0x418767e4)
000102.850|sip  |2|03|SipOnEvCallNewState 418767e4,42c7dd9c 7,Held
000102.907|sip  |1|03|MsgSipTcpPacket
000102.909|sip  |2|03|CCallBase::IsChallenged 'UPDATE' Dialog Tag '622959DD-88BF8770' pRequest Tag '622959DD-88BF8770' state 'Confirmed'
000102.909|sip  |2|03|new UA Server Non-INVITE trans state 'callingTrying', timeout=0 (0x41828fc4)
000102.909|sip  |3|03|GetRemotePartyAddress from 'P-Asserted-Identity'
Message 3 of 6
Highlighted
Polycom Employee & Community Manager

Re: Call held after answer

Hello Adrian,

The community's VoIP FAQ contains this post here:

Jan 19, 2012 Question: How to troubleshoot Polycom VoIP related Issues?

Resolution: Please check => here <=


The above shows how to get SIP debug logs as your log is not in debug.


Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

Polycom Global Services

----------------

Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge. If you need immediate and/or official assistance please open a service ticket through your proper support channels.
Please also ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's
Message 4 of 6
Highlighted
Occasional Advisor

Re: Call held after answer

PART ONE:

Here it is... the whole shabang with global debug on.  I did change the host names and IPs

 

I did upgrade since the last post to:

Phone Model     SoundStation IP 7000
Part Number     3111-40000-001 Rev:J
UC Software Version     4.0.10.0568
BootROM Software Version     5.0.1.10553

 

myhost.domain.com -> sip server

123.123.123.123 -> sip server

456.456.456.456 -> local gateway wan ip

10.10.10.1 -> internal address of sip server

10.10.10.50 -> internal address of '1111' phone (IP 7000) (Call from)

10.10.10.120 -> internal address of '2222' phone (VVX 500) (Call to)

 

000434.312|copy |4|03|Configuration of URL failed
000434.312|cfg  |4|03|Prov|Uploading phoneWeb.cfg failed
000435.713|sip  |0|03|>>> Data Send TCP port:5060
000435.713|sip  |0|03|    
000435.713|sip  |0|03|    
000435.713|sip  |0|03|    
000435.713|sip  |0|03|    
000435.713|sip  |0|03|    
000435.713|sip  |0|03|    
000435.713|sip  |0|03|    
000435.713|sip  |0|03|    
000435.736|sip  |0|03|<<< Data received TCP
000435.736|sip  |0|03|    
000435.736|sip  |1|03|MsgSipTcpPacket
000435.736|sip  |5|03|Can not decode the packet
000435.844|sip  |2|03|SipCallNew 0 local port 2224 call appearance -1 IsRtrv 0 dialog 0
000435.845|sip  |2|03|CStkDialog::CStkDialog SetAddressLocal Config '1111' <1111@host.domain.com:0>
000435.845|sip  |2|03|CStkDialog::CStkDialog AddressLocal set to Config
000435.845|sip  |3|03|CStkDialog::SetAddressLocal localTag set to ''
000435.845|sip  |3|03|CStkDialog::SetAddressLocal new address added of 1
000435.845|sip  |2|03|CStkDialog::CStkDialog TAG 'A26282A7-A7F800DC' generated
000435.845|sip  |2|03|CStkDialog::CStkDialog local addr '1111' <1111@host.domain.com:0> Tag 'A26282A7-A7F800DC'
000435.845|sip  |2|03|CStkDialog::CStkDialog exit 0x41938ff4 local list size 1
000435.845|sip  |2|03|CStkDialogList::CreateDialogObject localTarg usr '1111'
000435.845|sip  |2|03|CUser::CallNew 0x42d46e64 0x4192107c CallAppr 0 IsRetrieve 0 ThrdParty '' Dialog 0x0 isCentConf 0
000435.845|sip  |3|03|CStkCall::NewCallState reason 15 'Unknown'->'Dialtone' (0x4192107c)
000435.845|sip  |2|03|SipOnEvCallNewState 4192107c,42d46e64 0,Dialtone
000435.846|sip  |2|03|SipCallMake 2222
000435.846|sip  |2|03|new UA Client INVITE trans state 'callingTrying', timeout=0 (0x418cfcbc)
000435.846|sip  |1|03|CreateFailOverProxyList : Reg to Domain 'host.domain.com' nPort 0
000435.846|sip  |1|03|CreateFailOverProxyList : For INVITE Request nPort 0
000435.847|sip  |1|03|doDnsListLookup(tcp): doDnsSrvLookupForARecordList for 'host.domain.com' port 0 returned 1 results
000435.847|sip  |1|03|doDnsListLookup(tcp): result 0 '456.456.456.456' port 0 isInBound 0
000435.847|sip  |1|03|CreateFailOverProxyList : 'TCP Only' for 'host.domain.com' port 0 IP 0 is '456.456.456.456' on tcp port 0
000435.847|sip  |1|03|CreateFailOverProxyList : 'TCP Only' Add rest Total to Try 1
000435.847|sip  |2|03|CreateFailOverProxyList : Exit 'TCP Only' lookup with 1 IP Addresses
000435.847|sip  |2|03|CreateFailOverProxyList : IP 1 is '456.456.456.456' on tcp port 0
000435.847|sip  |1|03|CTcp::Send(TCP) address 456.456.456.456 port 5060 can Connect 1
000435.847|sip  |0|03|>>> Data Send TCP port:5060
000435.847|sip  |0|03|    INVITE sip:2222@host.domain.com;user=phone;transport=tcp SIP/2.0
000435.847|sip  |0|03|    Via: SIP/2.0/TCP 10.10.10.50;branch=z9hG4bKf62d6f9795B10EC
000435.847|sip  |0|03|    From: "1111" <sip:1111@host.domain.com>;tag=A26282A7-A7F800DC
000435.847|sip  |0|03|    To: <sip:2222@host.domain.com;user=phone>
000435.847|sip  |0|03|    CSeq: 1 INVITE
000435.847|sip  |0|03|    Call-ID: 1a76d287-658fcafc-1a846b77@10.10.10.50
000435.847|sip  |0|03|    Contact: <sip:1111@10.10.10.50;transport=tcp>
000435.847|sip  |0|03|    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
000435.847|sip  |0|03|    User-Agent: PolycomSoundStationIP-SSIP_7000-UA/4.0.10.0568
000435.847|sip  |0|03|    Accept-Language: en
000435.847|sip  |0|03|    Supported: 100rel,replaces
000435.847|sip  |0|03|    Allow-Events: conference,talk,hold
000435.847|sip  |0|03|    Max-Forwards: 70
000435.847|sip  |0|03|    Content-Type: application/sdp
000435.847|sip  |0|03|    Content-Length: 517
000435.847|sip  |0|03|    
000435.847|sip  |0|03|    v=0
000435.848|sip  |0|03|    o=- 1416037385 1416037385 IN IP4 10.10.10.50
000435.848|sip  |0|03|    s=Polycom IP Phone
000435.848|sip  |0|03|    c=IN IP4 10.10.10.50
000435.848|sip  |0|03|    t=0 0
000435.848|sip  |0|03|    a=sendrecv
000435.848|sip  |0|03|    m=audio 2224 RTP/AVP 106 115 99 9 102 0 8 18 127
000435.848|sip  |0|03|    a=rtpmap:106 SIREN22/48000
000435.848|sip  |0|03|    a=fmtp:106 bitrate=64000
000435.848|sip  |0|03|    a=rtpmap:115 G7221/32000
000435.848|sip  |0|03|    a=fmtp:115 bitrate=48000
000435.848|sip  |0|03|    a=rtpmap:99 SIREN14/16000
000435.848|sip  |0|03|    a=fmtp:99 bitrate=48000
000435.848|sip  |0|03|    a=rtpmap:9 G722/8000
000435.848|sip  |0|03|    a=rtpmap:102 G7221/16000
000435.848|sip  |0|03|    a=fmtp:102 bitrate=32000
000435.848|sip  |0|03|    a=rtpmap:0 PCMU/8000
000435.848|sip  |0|03|    a=rtpmap:8 PCMA/8000
000435.848|sip  |0|03|    a=rtpmap:18 G729/8000
000435.848|sip  |0|03|    a=fmtp:18 annexb=no
000435.848|sip  |0|03|    a=rtpmap:127 telephone-event/8000
000435.848|sip  |2|03|adjustRetransWhenTimerCreated UA Client INVITE INVITE state 'callingTrying' timeout=65 (0x418cfcbc)
000435.848|sip  |3|03|CStkCall::NewCallState reason 15 'Dialtone'->'Proceeding' (0x4192107c)
000435.849|sip  |2|03|SipOnEvCallNewState 4192107c,42d46e64 2,Proceeding
000435.875|sip  |0|03|<<< Data received TCP
000435.875|sip  |0|03|    SIP/2.0 100 Trying
000435.875|sip  |0|03|    Via: SIP/2.0/TCP 10.10.10.50;branch=z9hG4bKf62d6f9795B10EC;received=123.123.123.123;rport=61539
000435.875|sip  |0|03|    From: "1111" <sip:1111@host.domain.com>;tag=A26282A7-A7F800DC
000435.875|sip  |0|03|    To: <sip:2222@host.domain.com;user=phone>
000435.875|sip  |0|03|    Call-ID: 1a76d287-658fcafc-1a846b77@10.10.10.50
000435.875|sip  |0|03|    CSeq: 1 INVITE
000435.875|sip  |0|03|    User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20151023T152155Z~9fcf3e0c9a~64bit
000435.875|sip  |0|03|    Content-Length: 0
000435.875|sip  |0|03|    
000435.875|sip  |1|03|MsgSipTcpPacket
000435.876|sip  |1|03|SipOnCommand: response 100,INVITE
000435.876|sip  |1|03|SipOnCommand: response 100,INVITE matches user 1 of 1 '1111'
000435.876|sip  |3|03|UA Client INVITE INVITE trans state 'callingTrying'->'proceeding' by 100 resp 65 timeout(0x418cfcbc)
000435.876|sip  |2|03|CTrans:: INVITE InvTran reTrans ALREADY stopped in 'proceeding' state at retryCount 0 code 100, timeout=65 (0x418cfcbc)
000435.876|sip  |3|03|GetRemotePartyAddress from 'To'
000435.876|sip  |3|03|CStkCall::OnEvNewDest (0x4192107c) new display '' user '2222' old 'From' new 'To' source
000435.876|sip  |2|03|SipOnEvNewDest 4192107c,42d46e64,2222,
000435.877|sip  |3|03|CStkCall::NewCallState reason 15 'Proceeding'->'Proceeding' (0x4192107c)
000435.877|sip  |2|03|SipOnEvCallNewState 4192107c,42d46e64 2,Proceeding
000436.135|sip  |0|03|<<< Data received TCP
000436.136|sip  |0|03|    SIP/2.0 183 Session Progress
000436.136|sip  |0|03|    Via: SIP/2.0/TCP 10.10.10.50;branch=z9hG4bKf62d6f9795B10EC;received=123.123.123.123;rport=61539
000436.136|sip  |0|03|    From: "1111" <sip:1111@host.domain.com>;tag=A26282A7-A7F800DC
000436.136|sip  |0|03|    To: <sip:2222@host.domain.com;user=phone>;tag=j12K50Qj4Zc3a
000436.136|sip  |0|03|    Call-ID: 1a76d287-658fcafc-1a846b77@10.10.10.50
000436.136|sip  |0|03|    CSeq: 1 INVITE
000436.136|sip  |0|03|    Contact: <sip:2222@host.domain.com:5060;transport=tcp>
000436.136|sip  |0|03|    User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20151023T152155Z~9fcf3e0c9a~64bit
000436.136|sip  |0|03|    Accept: application/sdp
000436.136|sip  |0|03|    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
000436.136|sip  |0|03|    Supported: timer, path, replaces
000436.136|sip  |0|03|    Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
000436.136|sip  |0|03|    Content-Type: application/sdp
000436.136|sip  |0|03|    Content-Disposition: session
000436.136|sip  |0|03|    Content-Length: 230
000436.136|sip  |0|03|    Remote-Party-ID: "2222" <2222>;party=calling;privacy=off;screen=no
000436.136|sip  |0|03|    
000436.136|sip  |0|03|    v=0
000436.136|sip  |0|03|    o=FreeSWITCH 1460973605 1460973606 IN IP4 host.domain.com
000436.136|sip  |0|03|    s=FreeSWITCH
000436.136|sip  |0|03|    c=IN IP4 host.domain.com
000436.136|sip  |0|03|    t=0 0
000436.136|sip  |0|03|    m=audio 21936 RTP/AVP 9 127
000436.136|sip  |0|03|    a=rtpmap:9 G722/8000
000436.136|sip  |0|03|    a=rtpmap:127 telephone-event/8000
000436.136|sip  |0|03|    a=fmtp:127 0-16
000436.136|sip  |0|03|    a=ptime:20
000436.136|sip  |1|03|MsgSipTcpPacket
000436.137|sip  |1|03|SipOnCommand: response 183,INVITE
000436.137|sip  |1|03|SipOnCommand: response 183,INVITE matches user 1 of 1 '1111'
000436.137|sip  |3|03|CStateInviteServer::CStateInviteServer central conf user user '' found in contact user '2222' for cent conf URI ''. Set is focus
000436.137|sip  |3|03|CCallBase::OnEvResponse isFocus set for call 0x4192107c
000436.138|sip  |3|03|CStkDialog::CreateRouteSet: transport set to Target URI 'TCP'
000436.139|sip  |3|03|GetRemotePartyAddress from 'Remote-Party-ID'
000436.139|sip  |3|03|CStkCall::OnEvNewDest (0x4192107c) new display '2222' user '' old 'To' new 'Remote-Party-ID' source
000436.139|sip  |2|03|SipOnEvNewDest 4192107c,42d46e64,,2222
000436.139|sip  |1|03|Dialog 'id0dc05acc' State 'Trying'->'Early'
000436.139|sip  |3|03|CStkCall::NewCallState reason 15 'Proceeding'->'RingBack' (0x4192107c)
000436.139|sip  |2|03|SipOnEvCallNewState 4192107c,42d46e64 3,{NULL}
000436.140|sip  |3|03|CStkCall::RemoteSdpAnswer(1) -> ReportCodec( 0)
000436.140|sip  |2|03|CStkCall::ReportCodec: held set true due to m_MediaArray[i].m_inetAddr == 0
000436.140|sip  |2|03|SipOnEvNewCodec 0,9 9 G722/8000 21936,2224 ptime=20,dir 2 index 0 lastCodec 1 callWithVideo 0 bandwidth -1
Message 5 of 6
Highlighted
Occasional Advisor

Re: Call held after answer

PART 2:

 

000436.141|sip  |2|03|SipOnEvNewCodec 0,127 127 telephone-event/8000 21936,2224 ptime=0,dir 2 index 0 lastCodec 1 callWithVideo 0 bandwidth -1
000436.243|net  |2|03|NWIF: nw_setlocalhold() - setting local hold (0) 2.
000436.243|sip  |3|03|CStkCall::ReportCodec: call state 'RingBack' exit with held 1 (0x4192107c)
000436.243|sip  |3|03|CStateInviteClient::OnEvResponse 183 SDP present
000436.303|net  |2|03|NWIF: nw_setlocalhold() - setting local hold (0) 2.
000437.285|net  |4|03|rtosNetwork: netwTask() - Can't find associated CCB.
000437.498|net  |4|03|rtosNetwork net02: netwSend() - sendto() call failed. fd 1114832112 errno=130
000437.799|sip  |0|03|<<< Data received TCP
000437.799|sip  |0|03|    SIP/2.0 200 OK
000437.799|sip  |0|03|    Via: SIP/2.0/TCP 10.10.10.50;branch=z9hG4bKf62d6f9795B10EC;received=123.123.123.123;rport=61539
000437.799|sip  |0|03|    From: "1111" <sip:1111@host.domain.com>;tag=A26282A7-A7F800DC
000437.799|sip  |0|03|    To: <sip:2222@host.domain.com;user=phone>;tag=j12K50Qj4Zc3a
000437.799|sip  |0|03|    Call-ID: 1a76d287-658fcafc-1a846b77@10.10.10.50
000437.799|sip  |0|03|    CSeq: 1 INVITE
000437.799|sip  |0|03|    Contact: <sip:2222@host.domain.com:5060;transport=tcp>
000437.799|sip  |0|03|    User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20151023T152155Z~9fcf3e0c9a~64bit
000437.799|sip  |0|03|    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
000437.799|sip  |0|03|    Supported: timer, path, replaces
000437.799|sip  |0|03|    Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
000437.799|sip  |0|03|    Session-Expires: 120;refresher=uas
000437.799|sip  |0|03|    Content-Type: application/sdp
000437.799|sip  |0|03|    Content-Disposition: session
000437.799|sip  |0|03|    Content-Length: 230
000437.799|sip  |0|03|    Remote-Party-ID: "2222" <2222>;party=calling;privacy=off;screen=no
000437.799|sip  |0|03|    
000437.799|sip  |0|03|    v=0
000437.799|sip  |0|03|    o=FreeSWITCH 1460973605 1460973606 IN IP4 host.domain.com
000437.799|sip  |0|03|    s=FreeSWITCH
000437.799|sip  |0|03|    c=IN IP4 host.domain.com
000437.799|sip  |0|03|    t=0 0
000437.799|sip  |0|03|    m=audio 21936 RTP/AVP 9 127
000437.799|sip  |0|03|    a=rtpmap:9 G722/8000
000437.799|sip  |0|03|    a=rtpmap:127 telephone-event/8000
000437.799|sip  |0|03|    a=fmtp:127 0-16
000437.799|sip  |0|03|    a=ptime:20
000437.799|sip  |1|03|MsgSipTcpPacket
000437.801|sip  |1|03|SipOnCommand: response 200,INVITE
000437.801|sip  |1|03|SipOnCommand: response 200,INVITE matches user 1 of 1 '1111'
000437.801|sip  |3|03|CStkDialog::CreateRouteSet: transport set to Target URI 'TCP'
000437.801|sip  |3|03|UA Client INVITE INVITE trans state 'proceeding'->'terminated' by 200 resp 0 timeout(0x418cfcbc)
000437.801|sip  |1|03|CreateFailOverProxyList : Reg to Domain 'host.domain.com' nPort 5060
000437.801|sip  |1|03|CreateFailOverProxyList : For ACK Request nPort 5060
000437.801|sip  |1|03|doDnsListLookup(tcp): doDnsSrvLookupForARecordList for 'host.domain.com' port 5060 returned 1 results
000437.801|sip  |1|03|doDnsListLookup(tcp): result 0 '456.456.456.456' port 5060 isInBound 0
000437.801|sip  |1|03|CreateFailOverProxyList : 'TCP Only' for 'host.domain.com' port 5060 IP 0 is '456.456.456.456' on tcp port 5060
000437.801|sip  |2|03|CreateFailOverProxyList : Exit 'TCP Only' lookup with 1 IP Addresses
000437.802|sip  |2|03|CreateFailOverProxyList : IP 1 is '456.456.456.456' on tcp port 5060
000437.802|sip  |1|03|CTcp::Send(TCP) address 456.456.456.456 port 5060 can Connect 1
000437.802|sip  |0|03|>>> Data Send TCP port:5060
000437.802|sip  |0|03|    ACK sip:2222@host.domain.com:5060;transport=tcp SIP/2.0
000437.802|sip  |0|03|    Via: SIP/2.0/TCP 10.10.10.50;branch=z9hG4bK72a46f0c4AB3FA67
000437.802|sip  |0|03|    From: "1111" <sip:1111@host.domain.com>;tag=A26282A7-A7F800DC
000437.802|sip  |0|03|    To: <sip:2222@host.domain.com;user=phone>;tag=j12K50Qj4Zc3a
000437.802|sip  |0|03|    CSeq: 1 ACK
000437.802|sip  |0|03|    Call-ID: 1a76d287-658fcafc-1a846b77@10.10.10.50
000437.802|sip  |0|03|    Contact: <sip:1111@10.10.10.50;transport=tcp>
000437.802|sip  |0|03|    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
000437.802|sip  |0|03|    User-Agent: PolycomSoundStationIP-SSIP_7000-UA/4.0.10.0568
000437.802|sip  |0|03|    Accept-Language: en
000437.802|sip  |0|03|    Max-Forwards: 70
000437.802|sip  |0|03|    Content-Length: 0
000437.802|sip  |0|03|    
000437.802|sip  |2|03|adjustRetransWhenTimerCreated UA Client INVITE ACK state 'terminated' timeout=65 (0x418cfcbc)
000437.802|sip  |3|03|GetRemotePartyAddress from 'Remote-Party-ID'
000437.802|sip  |3|03|CStkCall::OnEvNewDest Unchanged display '2222' user ''
000437.802|sip  |2|03|CStateInviteClient::OnEvResponse Normal case
000437.803|sip  |3|03|CStkCall::RemoteSdpAnswer(1) -> ReportCodec( 1)
000437.803|sip  |2|03|CStkCall::ReportCodec: held set true due to m_MediaArray[i].m_inetAddr == 0
000437.803|sip  |2|03|SipOnEvNewCodec 0,9 9 G722/8000 21936,2224 ptime=20,dir 2 index 0 lastCodec 1 callWithVideo 0 bandwidth -1
000437.803|sip  |2|03|SipOnEvNewCodec 0,127 127 telephone-event/8000 21936,2224 ptime=0,dir 2 index 0 lastCodec 1 callWithVideo 0 bandwidth -1
000437.803|sip  |3|03|CStkCall::ReportCodec: call state 'RingBack' exit with held 1 (0x4192107c)
000437.803|sip  |1|03|Dialog 'id0dc05acc' State 'Early'->'Confirmed'
000437.803|sip  |3|03|CStkCall::NewCallState reason 15 'RingBack'->'Held' (0x4192107c)
000437.803|sip  |2|03|SipOnEvCallNewState 4192107c,42d46e64 7,Held
000437.804|net  |2|03|NWIF: nw_setlocalhold() - setting local hold (0) 2.
000437.858|sip  |0|03|<<< Data received TCP
000437.858|sip  |0|03|    UPDATE sip:1111@10.10.10.50;transport=tcp SIP/2.0
000437.858|sip  |0|03|    Via: SIP/2.0/TCP host.domain.com;rport;branch=z9hG4bK3cN2cr7aUXXpc
000437.858|sip  |0|03|    Max-Forwards: 70
000437.858|sip  |0|03|    From: <sip:2222@host.domain.com;user=phone>;tag=j12K50Qj4Zc3a
000437.858|sip  |0|03|    To: "1111" <sip:1111@host.domain.com>;tag=A26282A7-A7F800DC
000437.858|sip  |0|03|    Call-ID: 1a76d287-658fcafc-1a846b77@10.10.10.50
000437.858|sip  |0|03|    CSeq: 90161707 UPDATE
000437.858|sip  |0|03|    Contact: <sip:2222@host.domain.com:5060;transport=tcp>
000437.858|sip  |0|03|    User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20151023T152155Z~9fcf3e0c9a~64bit
000437.858|sip  |0|03|    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
000437.858|sip  |0|03|    Supported: timer, path, replaces
000437.858|sip  |0|03|    Session-Expires: 120;refresher=uac
000437.859|sip  |0|03|    Min-SE: 120
000437.859|sip  |0|03|    Content-Type: application/sdp
000437.859|sip  |0|03|    Content-Disposition: session
000437.859|sip  |0|03|    Content-Length: 230
000437.859|sip  |0|03|    P-Asserted-Identity: "2222" <sip:2222@host.domain.com>
000437.859|sip  |0|03|    
000437.859|sip  |0|03|    v=0
000437.859|sip  |0|03|    o=FreeSWITCH 1460973605 1460973606 IN IP4 host.domain.com
000437.859|sip  |0|03|    s=FreeSWITCH
000437.859|sip  |0|03|    c=IN IP4 host.domain.com
000437.859|sip  |0|03|    t=0 0
000437.859|sip  |0|03|    m=audio 21936 RTP/AVP 9 127
000437.859|sip  |0|03|    a=rtpmap:9 G722/8000
000437.859|sip  |0|03|    a=rtpmap:127 telephone-event/8000
000437.859|sip  |0|03|    a=fmtp:127 0-16
000437.859|sip  |0|03|    a=ptime:20
000437.859|sip  |1|03|MsgSipTcpPacket
000437.860|sip  |3|03|CStkDialog::IsThisDialog 'UPDATE' contact '<sip:2222@host.domain.com:5060;transport=tcp>' != '<sip:2222@host.domain.com:5060;transport=tcp>' caused a Dialog Target Refresh
000437.860|sip  |3|03|CStkDialog::CreateRouteSet: transport set to Target URI 'TCP'
000437.860|sip  |2|03|CCallBase::IsChallenged 'UPDATE' Dialog Tag 'A26282A7-A7F800DC' pRequest Tag 'A26282A7-A7F800DC' state 'Confirmed'
000437.860|sip  |2|03|new UA Server Non-INVITE trans state 'callingTrying', timeout=0 (0x418d365c)
000437.860|sip  |3|03|GetRemotePartyAddress from 'P-Asserted-Identity'
000437.860|sip  |3|03|CStkCall::OnEvNewDest (0x4192107c) new display '2222' user '2222' old 'Remote-Party-ID' new 'P-Asserted-Identity' source
000437.860|sip  |2|03|SipOnEvNewDest 4192107c,42d46e64,2222,2222
000437.861|sip  |2|03|Update comes with SDP
000437.862|sip  |3|03|UA Server Non-INVITE UPDATE trans state 'callingTrying'->'completed' by 200 resp 65 timeout(0x418d365c)
000437.862|sip  |1|03|doDnsListLookup(tcp): doDnsSrvLookupForARecordList for '456.456.456.456' port 5060 returned 1 results
000437.862|sip  |1|03|doDnsListLookup(tcp): result 0 '456.456.456.456' port 5060 isInBound 1
000437.862|sip  |1|03|doDnsListLookup(tcp): doDnsSrvLookupForARecordList for 'host.domain.com' port 0 returned 1 results
000437.862|sip  |1|03|doDnsListLookup(tcp): result 0 '456.456.456.456' port 0 isInBound 0
000437.862|sip  |1|03|CTcp::Send(TCP) address 456.456.456.456 port 5060 can Connect 1
000437.862|sip  |0|03|>>> Data Send TCP port:5060
000437.863|sip  |0|03|    SIP/2.0 200 OK
000437.863|sip  |0|03|    Via: SIP/2.0/TCP host.domain.com;rport;branch=z9hG4bK3cN2cr7aUXXpc
000437.863|sip  |0|03|    From: <sip:2222@host.domain.com;user=phone>;tag=j12K50Qj4Zc3a
000437.863|sip  |0|03|    To: "1111" <sip:1111@host.domain.com>;tag=A26282A7-A7F800DC
000437.863|sip  |0|03|    CSeq: 90161707 UPDATE
000437.863|sip  |0|03|    Call-ID: 1a76d287-658fcafc-1a846b77@10.10.10.50
000437.863|sip  |0|03|    Contact: <sip:1111@10.10.10.50;transport=tcp>
000437.863|sip  |0|03|    User-Agent: PolycomSoundStationIP-SSIP_7000-UA/4.0.10.0568
000437.863|sip  |0|03|    Accept-Language: en
000437.863|sip  |0|03|    Content-Type: application/sdp
000437.863|sip  |0|03|    Content-Length: 517
000437.863|sip  |0|03|    
000437.863|sip  |0|03|    v=0
000437.863|sip  |0|03|    o=- 1416037385 1416037385 IN IP4 10.10.10.50
000437.863|sip  |0|03|    s=Polycom IP Phone
000437.863|sip  |0|03|    c=IN IP4 10.10.10.50
000437.863|sip  |0|03|    t=0 0
000437.863|sip  |0|03|    a=sendrecv
000437.863|sip  |0|03|    m=audio 2224 RTP/AVP 106 115 99 9 102 0 8 18 127
000437.863|sip  |0|03|    a=rtpmap:106 SIREN22/48000
000437.863|sip  |0|03|    a=fmtp:106 bitrate=64000
000437.863|sip  |0|03|    a=rtpmap:115 G7221/32000
000437.863|sip  |0|03|    a=fmtp:115 bitrate=48000
000437.863|sip  |0|03|    a=rtpmap:99 SIREN14/16000
000437.863|sip  |0|03|    a=fmtp:99 bitrate=48000
000437.863|sip  |0|03|    a=rtpmap:9 G722/8000
000437.863|sip  |0|03|    a=rtpmap:102 G7221/16000
000437.863|sip  |0|03|    a=fmtp:102 bitrate=32000
000437.863|sip  |0|03|    a=rtpmap:0 PCMU/8000
000437.863|sip  |0|03|    a=rtpmap:8 PCMA/8000
000437.863|sip  |0|03|    a=rtpmap:18 G729/8000
000437.863|sip  |0|03|    a=fmtp:18 annexb=no
000437.863|sip  |0|03|    a=rtpmap:127 telephone-event/8000
000437.863|sip  |2|03|CTrans::InitRetrans for UA Server Non-INVITE UPDATE state 'completed' Server 2 of 2 (0x418d365c)
000437.863|sip  |2|03|Session audio comes
000439.100|copy |4|03|Configuration of URL failed
000439.101|log  |4|03|UtilLogC::uploadFifoLog: upload error. protocol 0 result = 8
000439.362|net  |4|03|rtosNetwork: netwTask() - Can't find associated CCB.
000441.286|net  |2|03|NWIF: nw_setlocalhold() - setting local hold (1) 2.
000441.322|sip  |2|03|SipCallDrop 4192107c,42d46e64 reason 6
000441.322|sip  |3|03|CStkCall::Drop(reason = 6) (0x4192107c)
000441.322|sip  |2|03|new UA Client Non-INVITE trans state 'callingTrying', timeout=0 (0x418d4e1c)
000441.323|sip  |1|03|CreateFailOverProxyList : Reg to Domain 'host.domain.com' nPort 5060
000441.323|sip  |1|03|CreateFailOverProxyList : For BYE Request nPort 5060
000441.323|sip  |1|03|doDnsListLookup(tcp): doDnsSrvLookupForARecordList for 'host.domain.com' port 5060 returned 1 results
000441.323|sip  |1|03|doDnsListLookup(tcp): result 0 '456.456.456.456' port 5060 isInBound 0
000441.323|sip  |1|03|CreateFailOverProxyList : 'TCP Only' for 'host.domain.com' port 5060 IP 0 is '456.456.456.456' on tcp port 5060
000441.323|sip  |2|03|CreateFailOverProxyList : Exit 'TCP Only' lookup with 1 IP Addresses
000441.323|sip  |2|03|CreateFailOverProxyList : IP 1 is '456.456.456.456' on tcp port 5060
000441.323|sip  |1|03|CTcp::Send(TCP) address 456.456.456.456 port 5060 can Connect 1
000441.323|sip  |0|03|>>> Data Send TCP port:5060
000441.323|sip  |0|03|    BYE sip:2222@host.domain.com:5060;transport=tcp SIP/2.0
000441.323|sip  |0|03|    Via: SIP/2.0/TCP 10.10.10.50;branch=z9hG4bK7a0f3d1c677B3F57
000441.323|sip  |0|03|    From: "1111" <sip:1111@host.domain.com>;tag=A26282A7-A7F800DC
000441.323|sip  |0|03|    To: <sip:2222@host.domain.com;user=phone>;tag=j12K50Qj4Zc3a
000441.323|sip  |0|03|    CSeq: 2 BYE
000441.323|sip  |0|03|    Call-ID: 1a76d287-658fcafc-1a846b77@10.10.10.50
000441.323|sip  |0|03|    Contact: <sip:1111@10.10.10.50;transport=tcp>
000441.323|sip  |0|03|    User-Agent: PolycomSoundStationIP-SSIP_7000-UA/4.0.10.0568
000441.323|sip  |0|03|    Accept-Language: en
000441.323|sip  |0|03|    Max-Forwards: 70
000441.323|sip  |0|03|    Content-Length: 0
000441.323|sip  |0|03|    
000441.324|sip  |1|03|Dialog 'id0dc05acc' State 'Confirmed'->'Terminated'
000441.324|sip  |3|03|CStkCall::NewCallState reason 15 'Held'->'Idle' (0x4192107c)
000441.324|sip  |2|03|SipOnEvCallNewState 4192107c,42d46e64 10,Idle
000441.366|sip  |0|03|<<< Data received TCP
000441.366|sip  |0|03|    SIP/2.0 200 OK
000441.366|sip  |0|03|    Via: SIP/2.0/TCP 10.10.10.50;branch=z9hG4bK7a0f3d1c677B3F57;received=123.123.123.123;rport=61539
000441.366|sip  |0|03|    From: "1111" <sip:1111@host.domain.com>;tag=A26282A7-A7F800DC
000441.366|sip  |0|03|    To: <sip:2222@host.domain.com;user=phone>;tag=j12K50Qj4Zc3a
000441.366|sip  |0|03|    Call-ID: 1a76d287-658fcafc-1a846b77@10.10.10.50
000441.366|sip  |0|03|    CSeq: 2 BYE
000441.366|sip  |0|03|    User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20151023T152155Z~9fcf3e0c9a~64bit
000441.366|sip  |0|03|    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
000441.366|sip  |0|03|    Supported: timer, path, replaces
000441.366|sip  |0|03|    Content-Length: 0
000441.366|sip  |0|03|    
000441.366|sip  |1|03|MsgSipTcpPacket
000441.367|sip  |1|03|SipOnCommand: response 200,BYE
000441.367|sip  |1|03|SipOnCommand: response 200,BYE matches user 1 of 1 '1111'
000441.367|sip  |3|03|UA Client Non-INVITE BYE trans state 'callingTrying'->'completed' by 200 resp 10 timeout(0x418d4e1c)
000441.367|sip  |2|03|CTrans:: BYE NonInv reTrans ALREADY stopped in 'completed' state at retryCount 0 code 200, timeout=10 (0x418d4e1c)
000441.367|sip  |2|03|CStateByeClient::OnEvResponse 200
000442.712|sip  |1|03|Client State finished ACK (0x41938ff4)
Message 6 of 6