Ringing issue

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jeffrey....@gmail.com

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Apr 22, 2019, 6:06:23 PM4/22/19
to sipxcom-users
I have a problem with phones ringing constantly when setup as a key system.  Specifically at one location.  Here is the setup

In bound call -> Attendant  --+--> Working Hours live attendant --> Extension with 11 phones  --> No answer, voice mail
                              |
                              +--> After hours play message

The report I get from my customers at that location is that the phones will ring randomly when a call comes in, and not all phones will ring.  The phones are Polycom VVX 500's, each configured with four line keys with one call per line key.  They are running 5.5.2 of the firmware.  My system is running sipXcom (18.12.20190228064135 2019-02-28EST06:02:57 localhost.localdomain) update 1.

I've considered downgrading the phones to an earlier version.

Any thoughts on what might be wrong?

- Jeff 

pmkr...@gmail.com

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Apr 23, 2019, 3:19:38 PM4/23/19
to sipxcom-users
Take a look at the PCAP for a problem call from Sipxcom - if you are seeing 11 invites to each of the phones being sent for an incoming call to that location, then the issue is network-related and some of the invites are not arriving to one or more phones.

Are you using VPN technology to connect the phones to Sipxcom at the problem location?

Peter

jeffrey....@gmail.com

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Apr 23, 2019, 6:15:52 PM4/23/19
to sipxcom-users
Thanks Peter,
All of the phones, phone servers and AudioCodes Mediant 1000 are all on the same network, same VLAN and no VPN's in the middle.  I'll fire up a PCAP and see what I can find.

jeffrey....@gmail.com

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Apr 23, 2019, 6:45:04 PM4/23/19
to sipxcom-users
Ran a PCAP and there are indeed 11 invites.  All the phones seem to acknowledge the call.  This is an older switch, so it may be worth my time to swap it out.


On Tuesday, April 23, 2019 at 2:19:38 PM UTC-5, pmkr...@gmail.com wrote:

Michael Picher

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Apr 23, 2019, 6:49:17 PM4/23/19
to Jeff Roberson, sipxcom-users
I wouldn't be afraid to try firmware 5.9.2...

Mike
--------------------------------------------------------
Michael W. Picher


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pmkr...@gmail.com

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Apr 23, 2019, 7:42:38 PM4/23/19
to sipxcom-users
The issue may be intermittent and only appears on certain times of the day or with particular traffic patterns. If you can, try and get the user community to report the issue when it happens and then look at the pcap and whether the invites/responses for 11 phones complete successfully. If you see dropouts consistently from some phones, then look at the phone log for errors - turn phone->sip-logging level to debug. Also keep an eye out for DNS errors in the phone logs and pcap.

jeffrey....@gmail.com

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Jun 11, 2019, 9:51:31 AM6/11/19
to sipxcom-users
This is still an ongoing issue.  It happens even when the phone system isn't busy.  I've replaced the network switches, upgraded to the latest Polycom firmware and still the ringing patterns are random.  In addition, when the call is placed on hold, it will not show up on all of the phones.  I was able to get a clean pcap of two phone calls.  

jeffrey....@gmail.com

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Jun 11, 2019, 10:48:58 AM6/11/19
to sipxcom-users
Looking at the SIP logs from one of the phones.  The call was still ringing yet there was a CANCEL message that came through.  Here is the log:

1182414.546|sip  |3|00|New Connection Accepted
1182414.546|sip  |0|00|Plcm Socket created 0x43002f38
1182414.547|sip  |1|00|Task name tTCPListen227
1182414.547|sip  |1|00|MsgSipTcpAccept
1182414.547|sip  |1|00|socket accepted 34500712
1182414.547|sip  |3|00|Waiting for a  New Connection 0x42703cb8 0x42703dc0
1182414.547|sip  |0|00|RecvThreadLocal : Invoking ListenForIncomingData()
1182414.547|sip  |3|00|Creating recv Thread
1182414.547|sip  |3|00|Waiting on recv Thread
1182414.547|sip  |0|00|<<< Data received TCP
1182414.547|sip  |0|00|    INVITE sip:87...@172.20.4.70;transport=tcp;x-sipX-nonat;callgroup=1924 SIP/2.0
1182414.547|sip  |0|00|    Record-Route: <sip:172.20.0.2:5060;lr;sipXecs-CallDest=AL%2CINT;sipXecs-rs=%2Aauth%7E.%2Afrom%7EMWMzNjgxODg1ODg%60.900_ntap%2Aid%7EOTM2NS05MTQ%60%2115d6b2268963f3e08e083c7885ce7bf5>
1182414.547|sip  |0|00|    Via: SIP/2.0/TCP 172.20.0.2;branch=z9hG4bK-XX-2c45BmKSdwx_v9RibkTfGXb5ww
1182414.547|sip  |0|00|    Via: SIP/2.0/TCP 172.20.0.2;branch=z9hG4bK-XX-2c36bM8gnCtbAJFUJilABnPd_A~vkb`ZU7Fe6PPrksIEnN0sQ;id=9365-914
1182414.547|sip  |0|00|    Via: SIP/2.0/TCP 172.20.0.2;branch=z9hG4bK-XX-2c25dp8x9EcsR6V28SvrCKqTTA~ylA2BaASs`BZOdeoBnOQZg
1182414.547|sip  |0|00|    Via: SIP/2.0/TCP 172.20.0.65:5060;branch=z9hG4bKac369211019;alias
1182414.547|sip  |0|00|    Max-Forwards: 16
1182414.547|sip  |0|00|    From: "ROBERS, JEFF" <sip:62055...@hutchgov.com>;tag=1c368188588
1182414.547|sip  |0|00|    To: <sip:19...@hutchgov.com;user=phone>
1182414.547|sip  |0|00|    Call-Id: 3679598871...@172.20.0.65
1182414.547|sip  |0|00|    Cseq: 1 INVITE
1182414.547|sip  |0|00|    Contact: <sip:6205551212@172.20.0.65:5060;transport=tcp;x-sipX-nonat>
1182414.547|sip  |0|00|    Supported: em,100rel,timer,replaces,path,resource-priority,sdp-anat
1182414.547|sip  |0|00|    Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
1182414.547|sip  |0|00|    User-Agent: Mediant 1000 - MSBG/v.6.60A.281.003
1182414.547|sip  |0|00|    Content-Type: application/sdp
1182414.547|sip  |0|00|    Content-Length: 250
1182414.547|sip  |0|00|    Date: Tue, 11 Jun 2019 14:40:31 GMT
1182414.547|sip  |0|00|    Expires: 240
1182414.547|sip  |0|00|    X-Sipx-Authidentity: <sip:19...@hutchgov.com;signature=5CFFBD5F%3Ab5399cb00d8cd272ec960f15d1226060>
1182414.547|sip  |0|00|    X-Sipx-Handled: X172.20.0.2-70.167.36.202
1182414.547|sip  |0|00|    
1182414.547|sip  |0|00|    v=0
1182414.547|sip  |0|00|    o=AudiocodesGW 366968663 366968634 IN IP4 172.20.0.65
1182414.547|sip  |0|00|    s=Phone-Call
1182414.547|sip  |0|00|    c=IN IP4 172.20.0.65
1182414.547|sip  |0|00|    t=0 0
1182414.547|sip  |0|00|    m=audio 8250 RTP/AVP 0 8 96
1182414.547|sip  |0|00|    a=ptime:20
1182414.548|sip  |0|00|    a=sendrecv
1182414.548|sip  |0|00|    a=rtpmap:0 PCMU/8000
1182414.548|sip  |0|00|    a=rtpmap:8 PCMA/8000
1182414.548|sip  |0|00|    a=rtpmap:96 telephone-event/8000
1182414.548|sip  |0|00|    a=fmtp:96 0-15
1182414.548|sip  |1|00|MsgSipTcpPacket
1182414.549|sip  |3|00|CStkDialog::CreateRouteSet: transport set to top route 'TCP'
1182414.549|sip  |3|00|CStkDialog::SetAddressLocal localTag set to ''
1182414.549|sip  |3|00|CStkDialog::SetAddressLocal new address added of 1
1182414.549|sip  |2|00|CStkDialog::CStkDialog SetAddressLocal from pRequest To: 'Animal Shelter' <19...@hutchgov.com:0>
1182414.550|sip  |2|00|CStkDialog::CStkDialog SetAddressLocal Config 'Animal Shelter' <87...@hutchgov.com:0>
1182414.550|sip  |2|00|CStkDialog::CStkDialog TAG 'F78C3693-75C14468' generated
1182414.550|sip  |2|00|CStkDialog::CStkDialog local addr 'Animal Shelter' <19...@hutchgov.com:0> Tag 'F78C3693-75C14468'
1182414.550|sip  |2|00|CStkDialog::CStkDialog exit 0x1bfd578 local list size 1
1182414.550|sip  |2|00|CLocalIPManager::GetLocalIP Local IP identified is 172.20.4.70 for Remote address 172.20.4.70
1182414.550|sip  |1|00|SetCallAppearance m_nCallIndex 0 nCallAppearance 0
1182414.550|sip  |2|00|CCallBase::IsChallenged COPIED Dialog Tag to pRequest 'F78C3693-75C14468'
1182414.550|sip  |2|00|CCallBase::IsChallenged 'INVITE' Dialog Tag 'F78C3693-75C14468' pRequest Tag 'F78C3693-75C14468' state 'Trying'
1182414.550|sip  |2|00|new UA Server INVITE trans state 'proceeding', timeout=0 (0x40f04468)
1182414.550|sip  |2|00|CStateInviteServer::CStateInviteServer() Entrance
1182414.550|sip  |2|00|CStateInviteServer::CStateInviteServerHandler m_bIsRemote:0 m_bFederationType:0 m_bIsBypassEnable:0
1182414.550|sip  |2|00|Bypass Disabled, Choosing Application SDP
1182414.551|sip  |2|00|CLocalIPManager::SelectMediaIP, Looking for IP for Addrtype: 2
1182414.551|sip  |2|00|CLocalIPManager::SelectMediaIP, Addresses: 172.20.4.70 Addrtype: 2
1182414.551|sip  |2|00|CStkCall::VoiceFmtpReadLine: Matched i=0, j=0, payload=0
1182414.551|sip  |2|00|CStkCall::VoiceFmtpReadLine: Matched i=1, j=1, payload=8
1182414.551|sip  |2|00|ParseVideoFmtp, newVideoCnt (0)
1182414.551|sip  |2|00|CLocalIPManager::SelectMediaIP, Looking for IP for Addrtype: 2
1182414.551|sip  |2|00|CLocalIPManager::SelectMediaIP, Addresses: 172.20.4.70 Addrtype: 2
1182414.551|sip  |2|00|CStkCall::VoiceFmtpReadLine: Matched i=0, j=0, payload=0
1182414.551|sip  |2|00|CStkCall::VoiceFmtpReadLine: Matched i=1, j=1, payload=8
1182414.551|sip  |2|00|CStateReInviteClient::SendResponse response Number = 100, nSubReason = -1
1182414.551|sip  |1|00|CStkDialog::SetDialogState: Dialog 'id04990361' State 'Trying'->'Early'
1182414.552|sip  |1|00|doDnsListLookup(tcp): doDnsSrvLookupForARecordList for '172.20.0.2' port 45502 returned 1 results
1182414.552|sip  |1|00|doDnsListLookup(tcp): result 0 '172.20.0.2' port 45502 isInBound 0
1182414.552|sip  |1|00|doDnsListLookup(tcp): doDnsSrvLookupForARecordList for '172.20.0.2' port 0 returned 1 results
1182414.552|sip  |1|00|doDnsListLookup(tcp): result 0 '172.20.0.2' port 0 isInBound 0
1182414.552|sip  |2|00|CUser::GetFailBackMode 'Timeout'
1182414.552|sip  |1|00|CTrans:: SendCommand | this=0x40f04468, bVQMonMessage=0, m_pCall->m_pUser->m_bOBFailOverReRegOn=0, m_pCall->m_pUser->m_bVQMonFailoverEnabled=1
1182414.552|sip  |1|00|Send: (TCP) entry for address 172.20.0.2 port 45502 can Connect 0 canFailOver 1
1182414.552|sip  |1|00|Send: (TCP) address 172.20.0.2 port 45502 can Connect 0
1182414.552|sip  |0|00|IsThisSocket : Found Socket (227)
1182414.552|sip  |0|00|Send : Found Socket (227) 
1182414.553|sip  |0|00|TCP Data Send to 172.20.0.2:45502
1182414.553|sip  |0|00|    SIP/2.0 100 Trying
1182414.553|sip  |0|00|    Via: SIP/2.0/TCP 172.20.0.2;branch=z9hG4bK-XX-2c45BmKSdwx_v9RibkTfGXb5ww
1182414.553|sip  |0|00|    Via: SIP/2.0/TCP 172.20.0.2;branch=z9hG4bK-XX-2c36bM8gnCtbAJFUJilABnPd_A~vkb`ZU7Fe6PPrksIEnN0sQ;id=9365-914
1182414.553|sip  |0|00|    Via: SIP/2.0/TCP 172.20.0.2;branch=z9hG4bK-XX-2c25dp8x9EcsR6V28SvrCKqTTA~ylA2BaASs`BZOdeoBnOQZg
1182414.553|sip  |0|00|    Via: SIP/2.0/TCP 172.20.0.65:5060;branch=z9hG4bKac369211019;alias
1182414.553|sip  |0|00|    From: "ROBERS, JEFF" <sip:62055...@hutchgov.com>;tag=1c368188588
1182414.553|sip  |0|00|    To: "Animal Shelter" <sip:19...@hutchgov.com;user=phone>;tag=F78C3693-75C14468
1182414.553|sip  |0|00|    CSeq: 1 INVITE
1182414.553|sip  |0|00|    Call-ID: 3679598871...@172.20.0.65
1182414.553|sip  |0|00|    Contact: <sip:87...@172.20.4.70;transport=tcp>
1182414.553|sip  |0|00|    Record-Route: <sip:172.20.0.2:5060;lr;sipXecs-CallDest=AL%2CINT;sipXecs-rs=%2Aauth%7E.%2Afrom%7EMWMzNjgxODg1ODg%60.900_ntap%2Aid%7EOTM2NS05MTQ%60%2115d6b2268963f3e08e083c7885ce7bf5>
1182414.553|sip  |0|00|    User-Agent: PolycomVVX-VVX_500-UA/5.9.2.3446
1182414.553|sip  |0|00|    Accept-Language: en
1182414.553|sip  |0|00|    Content-Length: 0
1182414.553|sip  |0|00|    
1182414.553|sip  |0|00| Send : >>> End of data send
1182414.553|sip  |1|00|CPlcmSipTcpSocket::Send TCP queuedTxData = 0 TotalLen 892 loop count 1 maxQueueDepth 40000
1182414.553|sip  |3|00|CStkCall::NewCallState 'Unknown'->'Offering' (0x1bd2c38) m_hUI((nil)),Control Channel(0), ResponseCode(-1)
1182414.554|sip  |3|00|Use common source preference for incoming and outgoing calls
1182414.554|sip  |0|00|[CCommand::NeedToProcessCID] cmdType = 0 cmdMessage = 0 g_csSipRequestSourceMessage = -1 g_csSipResponseSourceMessage = -1---
1182414.554|sip  |3|00|GetRemotePartyAddress from 'From'
1182414.554|sip  |2|00|CStkCall::OnEvSubmitDest CallIdType(1)
1182414.554|cfg  |5|00|Prm|Array parameter reg.x.telephony index 269 is out of range
1182414.555|cfg  |5|00|Prm|2DArray parameter reg.x.server.x.specialInterop index 269 1 is out of range
1182414.555|cfg  |5|00|Prm|2DArray parameter reg.x.server.x.specialInterop index 269 2 is out of range
1182414.555|cfg  |5|00|Prm|2DArray parameter reg.x.server.x.specialInterop index 269 1 is out of range
1182414.558|sip  |3|00|Waiting on recv Thread
1182414.600|sip  |0|00|<<< Data received TCP
1182414.600|sip  |0|00|    CANCEL sip:87...@172.20.4.70;transport=tcp;x-sipX-nonat;callgroup=1924 SIP/2.0
1182414.600|sip  |0|00|    From: "ROBERS, JEFF" <sip:62055...@hutchgov.com>;tag=1c368188588
1182414.600|sip  |0|00|    To: <sip:19...@hutchgov.com;user=phone>
1182414.600|sip  |0|00|    Call-Id: 3679598871...@172.20.0.65
1182414.600|sip  |0|00|    Cseq: 1 CANCEL
1182414.600|sip  |0|00|    Max-Forwards: 20
1182414.600|sip  |0|00|    Via: SIP/2.0/TCP 172.20.0.2;branch=z9hG4bK-XX-2c45BmKSdwx_v9RibkTfGXb5ww
1182414.600|sip  |0|00|    Content-Length: 0
1182414.600|sip  |0|00|    
1182414.600|sip  |3|00|Waiting on recv Thread
1182414.605|sip  |1|00|SetCallAppearance m_nCallIndex 1 nCallAppearance 1
1182414.605|sip  |2|00|SipOnEvCallNewState 0x1bd2c38,0x1e513e0 1,(null), ResponseCode:-1
1182414.611|sip  |1|00|[MS-CALL-PARK]: NO PARK OPTION for 0x1bd2c38
1182414.611|sip  |2|00|CStkCall::ExtractAocBodyAndSend: IMS/AOC not enabled
1182414.611|sip  |1|00|MsgSipTcpPacket
1182414.612|sip  |2|00|active dialog after UPDATE being set here, m_pActiveDialog=0x1bfd578,this=0x1bfd578
1182414.612|sip  |2|00|new UA Server Non-INVITE trans state 'callingTrying', timeout=0 (0x40f077e8)
1182414.612|sip  |3|00|UA Server Non-INVITE CANCEL trans state 'callingTrying'->'completed' by 200 resp 65 timeout(0x40f077e8)
1182414.612|sip  |1|00|CStkDialog::SetDialogState: Dialog 'id04990361' State 'Early'->'Terminated'
1182414.612|sip  |1|00|doDnsListLookup(tcp): doDnsSrvLookupForARecordList for '172.20.0.2' port 45502 returned 1 results
1182414.612|sip  |1|00|doDnsListLookup(tcp): result 0 '172.20.0.2' port 45502 isInBound 0
1182414.613|sip  |1|00|doDnsListLookup(tcp): doDnsSrvLookupForARecordList for '172.20.0.2' port 0 returned 1 results
1182414.613|sip  |1|00|doDnsListLookup(tcp): result 0 '172.20.0.2' port 0 isInBound 0
1182414.613|sip  |2|00|CUser::GetFailBackMode 'Timeout'
1182414.613|sip  |1|00|CTrans:: SendCommand | this=0x40f077e8, bVQMonMessage=0, m_pCall->m_pUser->m_bOBFailOverReRegOn=0, m_pCall->m_pUser->m_bVQMonFailoverEnabled=1
1182414.614|sip  |1|00|Send: (TCP) entry for address 172.20.0.2 port 45502 can Connect 0 canFailOver 1
1182414.614|sip  |1|00|Send: (TCP) address 172.20.0.2 port 45502 can Connect 0
1182414.614|sip  |0|00|IsThisSocket : Found Socket (227)
1182414.614|sip  |0|00|Send : Found Socket (227) 
1182414.615|sip  |0|00|TCP Data Send to 172.20.0.2:45502
1182414.615|sip  |0|00|    SIP/2.0 200 OK
1182414.615|sip  |0|00|    Via: SIP/2.0/TCP 172.20.0.2;branch=z9hG4bK-XX-2c45BmKSdwx_v9RibkTfGXb5ww
1182414.615|sip  |0|00|    From: "ROBERS, JEFF" <sip:62055...@hutchgov.com>;tag=1c368188588
1182414.615|sip  |0|00|    To: "Animal Shelter" <sip:19...@hutchgov.com;user=phone>
1182414.615|sip  |0|00|    CSeq: 1 CANCEL
1182414.615|sip  |0|00|    Call-ID: 3679598871...@172.20.0.65
1182414.615|sip  |0|00|    Contact: <sip:87...@172.20.4.70;transport=tcp>
1182414.615|sip  |0|00|    User-Agent: PolycomVVX-VVX_500-UA/5.9.2.3446
1182414.615|sip  |0|00|    Accept-Language: en
1182414.615|sip  |0|00|    Content-Length: 0
1182414.615|sip  |0|00|    
1182414.615|sip  |0|00| Send : >>> End of data send
1182414.615|sip  |1|00|CPlcmSipTcpSocket::Send TCP queuedTxData = 0 TotalLen 410 loop count 1 maxQueueDepth 40000
1182414.615|sip  |2|00|CTrans::InitRetrans for UA Server Non-INVITE CANCEL state 'completed' Server 2 of 2 (0x40f077e8)

pmkr...@gmail.com

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Jun 11, 2019, 5:08:30 PM6/11/19
to sipxcom-users
If the CANCEL was being generated from the Audiocodes gateway, I would have expected another VIA record in the CANCEL packet pointing to the gateway. I'd look at the Sipxproxy log in INFO mode to see where the CANCEL is being generated - it may one of the extensions in the 1924 hunt group.

I assume from your comment that phones not displaying the red flashing light when placed on hold suggests that the line is a shared line. Do an audit on the phones with the shared line to ascertain that SIP->SIP SERVER 1 address or LINES->OUTBOUND Porxy address is provisioned to hutchgov.com. These fields are provisioned in the Phone SIP Server field or Line->Registrations->Outbound proxy fields in Sipxcom.

Peter

On Tuesday, June 11, 2019 at 10:48:58 AM UTC-4, jeffrey...@gmail.com wrote:
Looking at the SIP logs from one of the phones.  The call was still ringing yet there was a CANCEL message that came through.  Here is the log:

1182414.546|sip  |3|00|New Connection Accepted
1182414.546|sip  |0|00|Plcm Socket created 0x43002f38
1182414.547|sip  |1|00|Task name tTCPListen227
1182414.547|sip  |1|00|MsgSipTcpAccept
1182414.547|sip  |1|00|socket accepted 34500712
1182414.547|sip  |3|00|Waiting for a  New Connection 0x42703cb8 0x42703dc0
1182414.547|sip  |0|00|RecvThreadLocal : Invoking ListenForIncomingData()
1182414.547|sip  |3|00|Creating recv Thread
1182414.547|sip  |3|00|Waiting on recv Thread
1182414.547|sip  |0|00|<<< Data received TCP
1182414.547|sip  |0|00|    INVITE sip:...@172.20.4.70;transport=tcp;x-sipX-nonat;callgroup=1924 SIP/2.0
1182414.547|sip  |0|00|    Record-Route: <sip:172.20.0.2:5060;lr;sipXecs-CallDest=AL%2CINT;sipXecs-rs=%2Aauth%7E.%2Afrom%7EMWMzNjgxODg1ODg%60.900_ntap%2Aid%7EOTM2NS05MTQ%60%2115d6b2268963f3e08e083c7885ce7bf5>
1182414.547|sip  |0|00|    Via: SIP/2.0/TCP 172.20.0.2;branch=z9hG4bK-XX-2c45BmKSdwx_v9RibkTfGXb5ww
1182414.547|sip  |0|00|    Via: SIP/2.0/TCP 172.20.0.2;branch=z9hG4bK-XX-2c36bM8gnCtbAJFUJilABnPd_A~vkb`ZU7Fe6PPrksIEnN0sQ;id=9365-914
1182414.547|sip  |0|00|    Via: SIP/2.0/TCP 172.20.0.2;branch=z9hG4bK-XX-2c25dp8x9EcsR6V28SvrCKqTTA~ylA2BaASs`BZOdeoBnOQZg
1182414.547|sip  |0|00|    Via: SIP/2.0/TCP 172.20.0.65:5060;branch=z9hG4bKac369211019;alias
1182414.547|sip  |0|00|    Max-Forwards: 16
1182414.547|sip  |0|00|    From: "ROBERS, JEFF" <sip:620...@hutchgov.com>;tag=1c368188588
1182414.547|sip  |0|00|    To: <sip:...@hutchgov.com;user=phone>
1182414.547|sip  |0|00|    Call-Id: 3679598871...@172.20.0.65
1182414.547|sip  |0|00|    Cseq: 1 INVITE
1182414.547|sip  |0|00|    Contact: <sip:62055...@172.20.0.65:5060;transport=tcp;x-sipX-nonat>
1182414.547|sip  |0|00|    Supported: em,100rel,timer,replaces,path,resource-priority,sdp-anat
1182414.547|sip  |0|00|    Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
1182414.547|sip  |0|00|    User-Agent: Mediant 1000 - MSBG/v.6.60A.281.003
1182414.547|sip  |0|00|    Content-Type: application/sdp
1182414.547|sip  |0|00|    Content-Length: 250
1182414.547|sip  |0|00|    Date: Tue, 11 Jun 2019 14:40:31 GMT
1182414.547|sip  |0|00|    Expires: 240
1182414.547|sip  |0|00|    X-Sipx-Authidentity: <sip:...@hutchgov.com;signature=5CFFBD5F%3Ab5399cb00d8cd272ec960f15d1226060>
1182414.553|sip  |0|00|    From: "ROBERS, JEFF" <sip:620...@hutchgov.com>;tag=1c368188588
1182414.553|sip  |0|00|    To: "Animal Shelter" <sip:...@hutchgov.com;user=phone>;tag=F78C3693-75C14468
1182414.553|sip  |0|00|    CSeq: 1 INVITE
1182414.553|sip  |0|00|    Call-ID: 3679598871...@172.20.0.65
1182414.553|sip  |0|00|    Contact: <sip:...@172.20.4.70;transport=tcp>
1182414.553|sip  |0|00|    Record-Route: <sip:172.20.0.2:5060;lr;sipXecs-CallDest=AL%2CINT;sipXecs-rs=%2Aauth%7E.%2Afrom%7EMWMzNjgxODg1ODg%60.900_ntap%2Aid%7EOTM2NS05MTQ%60%2115d6b2268963f3e08e083c7885ce7bf5>
1182414.553|sip  |0|00|    User-Agent: PolycomVVX-VVX_500-UA/5.9.2.3446
1182414.553|sip  |0|00|    Accept-Language: en
1182414.553|sip  |0|00|    Content-Length: 0
1182414.553|sip  |0|00|    
1182414.553|sip  |0|00| Send : >>> End of data send
1182414.553|sip  |1|00|CPlcmSipTcpSocket::Send TCP queuedTxData = 0 TotalLen 892 loop count 1 maxQueueDepth 40000
1182414.553|sip  |3|00|CStkCall::NewCallState 'Unknown'->'Offering' (0x1bd2c38) m_hUI((nil)),Control Channel(0), ResponseCode(-1)
1182414.554|sip  |3|00|Use common source preference for incoming and outgoing calls
1182414.554|sip  |0|00|[CCommand::NeedToProcessCID] cmdType = 0 cmdMessage = 0 g_csSipRequestSourceMessage = -1 g_csSipResponseSourceMessage = -1---
1182414.554|sip  |3|00|GetRemotePartyAddress from 'From'
1182414.554|sip  |2|00|CStkCall::OnEvSubmitDest CallIdType(1)
1182414.554|cfg  |5|00|Prm|Array parameter reg.x.telephony index 269 is out of range
1182414.555|cfg  |5|00|Prm|2DArray parameter reg.x.server.x.specialInterop index 269 1 is out of range
1182414.555|cfg  |5|00|Prm|2DArray parameter reg.x.server.x.specialInterop index 269 2 is out of range
1182414.555|cfg  |5|00|Prm|2DArray parameter reg.x.server.x.specialInterop index 269 1 is out of range
1182414.558|sip  |3|00|Waiting on recv Thread
1182414.600|sip  |0|00|<<< Data received TCP
1182414.600|sip  |0|00|    CANCEL sip:...@172.20.4.70;transport=tcp;x-sipX-nonat;callgroup=1924 SIP/2.0
1182414.600|sip  |0|00|    From: "ROBERS, JEFF" <sip:620...@hutchgov.com>;tag=1c368188588
1182414.600|sip  |0|00|    To: <sip:...@hutchgov.com;user=phone>
1182414.615|sip  |0|00|    From: "ROBERS, JEFF" <sip:620...@hutchgov.com>;tag=1c368188588
1182414.615|sip  |0|00|    To: "Animal Shelter" <sip:...@hutchgov.com;user=phone>
1182414.615|sip  |0|00|    CSeq: 1 CANCEL
1182414.615|sip  |0|00|    Call-ID: 3679598871...@172.20.0.65
1182414.615|sip  |0|00|    Contact: <sip:...@172.20.4.70;transport=tcp>

jeffrey....@gmail.com

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Jun 14, 2019, 4:57:22 PM6/14/19
to sipxcom-users
There was an IP address for the sipXcom server listed for the outbound proxy.  I removed that from all of the phones, sent new profiles.  Similar result.

I then removed all of the phones from the system, removing any traces from the tftp server as well.  I then formatted each phone.  Let it auto configure.  Then I added the lines back to each phone.

Same problem.

I'm wondering if there is a configuration issue with the gateway, a Mediant 1000 running 6.60A.281.003.  
I'm seeing the call on the phones log and coming from the Mediants address.  

pmkr...@gmail.com

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Jun 14, 2019, 6:00:16 PM6/14/19
to sipxcom-users
Hi Jeff,

Let's separate the two issues. First ascertain that your shared line configuration is configured correctly - follow the steps outlined here http://wiki.ezuce.com/display/sipXcom/Setting+Up+Bridge+Line+Appearance+Lines+for+Small+and+Medium+Businesses.

Test your shared lines first with an internal call to a shared line to ascertain red flashing lamps appear on shared lines when the call is placed on hold. If your configuration looks good and shared lines do not work correctly, restart the sipxrls and sipxsaa processes. When you get everything working for an internal call, then place an external incoming call into a shared line. If the lights do not work properly, then I suspect there is a configuration issue on the Mediant gateway.

Back to your initial issue of not all phones ringing, check your firewall by doing an iptables -L command to ascertain there are no phones in the banned list.

Hope this helps.

Peter

jeffrey....@gmail.com

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Jun 17, 2019, 4:22:21 PM6/17/19
to sipxcom-users

Thanks Peter,

I went back through the doc, line-by-line to make sure I didn't miss something.  I did not setup the outbound proxy for the lines when I set the phones back up again.  I'll see if the users report any change.  Back to basics!

- Jeff

ivar....@onrelay.net

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Jun 24, 2019, 4:55:05 PM6/24/19
to sipxcom-users
Check if SIP ALG is enabled on the customer firewall, seemingly random incoming call behavior like this is a telltale sign. Note SIP ALG goes by many different names with the router manufacturers.

jeffrey....@gmail.com

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Jun 24, 2019, 7:05:40 PM6/24/19
to sipxcom-users

I want into the firewall (a SonicWall) and checked those settings.  Apparently they were enabled.  They are disabled now, I'll see if things change at all.  Thank you for the tip!
They are still having inconstant ringing issues with the phones, they receive mostly outside calls.

ivar....@onrelay.net

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Jun 24, 2019, 7:11:04 PM6/24/19
to sipxcom-users
We have battled SonicWalls at a few customers. Consistent NAT must be set to ON and SIP Transformation (their SIP ALG) set to OFF. You must probably also reboot the Polycoms for changes to take effect.

jeffrey....@gmail.com

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Jun 24, 2019, 8:07:33 PM6/24/19
to sipxcom-users
I did send out profiles to all of the phones. I’m not sure how much the SonicWall is actually in the loop with this issue. The AudoCodes Mediant, sipXcom system and all of the Polycom phones are all on the same VLAN, with the same address space. There is a random phone here and there that might be on the wrong VLAN that would route through the SonicWall.
This has been very frustrating for this building. This has been a long term issue for them.

ivar....@onrelay.net

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Jun 24, 2019, 8:17:35 PM6/24/19
to sipxcom-users
You are right these settings only affect SIP traffic across router NAT interfaces, which doesn't happen if everything is on the same VLAN.
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