VVX 600 Unable To Call Out on SIP Line

SOLVED
Occasional Advisor

VVX 600 Unable To Call Out on SIP Line

I am attempting to configure Line 1 of a Polycom VVX 600 (software v5.9.1.0615) to use an SIP line hosted by eTollFree.net.  I am able to receive both internal (i.e., extension-to-extension) and external calls on the Polycom, and am able to make internal calls, but I am unable to make external 11-digit calls (1 + area code + number) on the Polycom.  When I configure X-Lite to use the same extension, it can make internal and external calls without issue, so there must be something I am doing wrong when configuring the Polycom.  I attached a screenshot of what the Line 1 configuration screen looks like.  And here is the log file showing events registered when I saved the Line 1 settings and then attempted to dial a 11-digit and 10-digit (omitted the “1”) phone number:

0207102752|so   |4|00|[SoNcasC]: appncascontext termination:1
0207102752|so   |4|00|[SoNcasC]: Case Handling termination:1
0207102752|sip  |*|00|Sip UnRegister Usr:8203 Dsp:3123006782 Auth:'8203' Inx:1
0207102752|sip  |4|00|SipRemoveMonitoredUser : CSTA Line Not Found 
0207102752|sip  |*|00|SipUserRemove: user 1 being removed.
0207102752|so   |4|00|[SoNcasC]: appncascontext termination:1
0207102752|so   |4|00|[SoNcasC]: Case Handling termination:1
0207102752|app1 |*|00|SoRegistrationEventLineChanged - success lineIndex 0 RegListSize 0
0207102752|app1 |5|00|AppPhoneLockC::Init - bPhoneLockState [0]
0207102752|app1 |*|00|SoRegistrationEventLast - new AppRegLineC, szUser = 8201
0207102752|so   |4|00|[SoNcasC]: appncascontext termination:1
0207102752|so   |4|00|[SoNcasC]: Case Handling termination:1
0207102752|app1 |*|00|SoRegistrationEventLast - new AppRegLineC, szUser = 8203
0207102752|so   |4|00|[SoNcasC]: appncascontext termination:1
0207102752|so   |4|00|[SoNcasC]: Case Handling termination:1
0207102752|sip  |*|00|Sip UnRegister Usr:8201 Dsp:3123006782 Auth:'8201' Inx:0
0207102752|sip  |4|00|SipRemoveMonitoredUser : CSTA Line Not Found 
0207102752|sip  |*|00|SipUserRemove: user 0 being removed.
0207102752|cfg  |5|00|Prm|Parameter reg.x.outboundProxy.port requested type 0 but is of type 2
0207102752|sip  |*|00|Sip Register Usr:8201 Dsp:3123006782 Auth:'8201' Inx:0
0207102752|utilm|4|00|uBLFUnCompressed: File /ffs0/Config/Local/WebTicket/0/sip.usr doesn't exist or is empty
0207102752|cfg  |5|00|Prm|Parameter reg.x.outboundProxy.port requested type 0 but is of type 2
0207102752|sip  |*|00|Sip Register Usr:8203 Dsp:3123006782 Auth:'8203' Inx:1
0207102752|utilm|4|00|uBLFUnCompressed: File /ffs0/Config/Local/WebTicket/0/sip.usr doesn't exist or is empty
0207102752|app1 |5|00|AppPhoneLockC::Init - bPhoneLockState [0]
0207102756|sip  |*|00|User removed
0207102756|sip  |*|00|User removed
0207102802|copy |4|00|Configuration of URL failed
0207102802|cfg  |4|00|Prov|Could not download file 0004f2b06cc4-web.cfg
0207102802|cfg  |4|00|Prov|Uploading phoneWeb.cfg failed
0207102802|cfg  |4|00|Prov|Update configuration failed
0207102906|clist|4|00|dbCfg::getServerDir:Unknown dbCfg type
0207102906|clist|4|00|dbCfg::getServerDir:Unknown dbCfg type
0207102937|copy |4|00|Configuration of URL failed
0207102937|clist|4|00|dbIO::processResult:no host
0207103151|clist|4|00|dbCfg::getServerDir:Unknown dbCfg type
0207103151|clist|4|00|dbCfg::getServerDir:Unknown dbCfg type
0207103237|copy |4|00|Configuration of URL failed
0207103237|clist|4|00|dbIO::processResult:no host

I'm sure the answer is in that log file somewhere but I'm not sure what to fix.  Any help greatly appreciated!

9 REPLIES
Polycom Employee & Community Manager

Re: VVX 600 Unable To Call Out on SIP Line

Hello @gurs ,

 

welcome to the Polycom Community.

 

NETXUSA sold this phone back in 18/04/2013 so we assume this is a 2nd hand purchase as it must have worked at some point in its life.

 

The logs shows nothing but most likely this is a digitmap issue:

 

Oct 7, 2011 Question: Phone unable to Dial a number when Off Hook or on 2nd Call in a Conference or Digitmap issues

Resolution: Please check => here <=

 

and

 

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

Resolution: Please check => here <=

 

Next reply should at least contain a backup of the phone and valid logs.


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

Please be aware:

The purpose of these forums is to allow community members collaborate and help each other.
Questions posted here do not follow Polycom’s SLA guidelines.
If you require assistance from Polycom technical support, please open a
web service request or call us .

The above is necessary in order to track issue internally within Polycom.

You are welcome to post more questions or configuration or logs for other community members to look at but if your issue requires a fix via Polycom you must go via the official support structure.

Please ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's

Please remember, if you see a post that helped you , and it answers your question, please mark it as an "Accept as Solution".

This forum reply or post is based upon my personal experience and does not reflect the opinion or view of my employer.
Polycom employee participation within this community is not mandatory and any post or FAQ article provided by myself is done either during my working hours or outside working hours, in my private time, and may be answered on weekends, bank holidays or personal holidays.
Message 2 of 10
Occasional Advisor

Re: VVX 600 Unable To Call Out on SIP Line

@SteffenBaierUK, we actually purchased this phone new from Nextiva in 2013 when we were setting up service with them.  I have since moved our hosting to eTollFree and am trying to get a few of these phones to work with the new service.  I thought that would be a no-brainer, but apparently not!  Thanks for the tips, I will troubleshoot when I am back in front of the device and report back.

Message 3 of 10
Occasional Advisor

Re: VVX 600 Unable To Call Out on SIP Line

@SteffenBaierUK, I changed the digitmap to match the one in the link you provided.  My starting digitmap was:

[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT|**x.T|+x.T

Which I changed to:

[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT

This change did not solve my problem.  I also checked the second link you suggested regarding troubleshooting Polycom VoIP-related issues.  I confirmed that my Polycom log settings matched the recommendations.  I then rebooted the VVX 600 and attempted to dial an 11-digit number and a 10-digit number.  Both attempts timed out after about 30 seconds with a fast busy signal.  I then attempted to dial a 4-digit extension, which went through without issue.  I should also point out that a softphone running on a PC that is on the same LAN as the Polycom is able to send/receive calls using the same SIP account, so it seems unlikely that this is a LAN/router issue.

 

I have attached a zip archive containing the log file and phone backup.  Please advise of next steps.  Thanks again for your help - I am in way over my head on this one!

Polycom Employee & Community Manager

Re: VVX 600 Unable To Call Out on SIP Line

Hello @gurs ,

 

this landed in the SPAM so I moved this for you.

 

You have not followed the FAQ's so I got limited information I can provide.

 

Please also remember that this is a open Forum with volunteers answering.

 

You called:

0208142626|sip  |1|00|[CInvite]: szDest  - 13129445852

And we tried to send this but the server never responded:

	Line 1735: 0208142626|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send   500 of max 31500
	Line 1739: 0208142627|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send  1500 of max 31500
	Line 1743: 0208142629|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send  3500 of max 31500
	Line 1749: 0208142633|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send  7500 of max 31500
	Line 1753: 0208142641|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send 15500 of max 31500
	Line 1757: 0208142657|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send 31500 of max 31500

You then called

0208142707|sip  |1|00|[CInvite]: szDest  - 3129445852

And we tried to send this but the server never responded:

	Line 1863: 0208142708|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send   500 of max 31500
	Line 1867: 0208142709|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send  1500 of max 31500
	Line 1871: 0208142711|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send  3500 of max 31500
	Line 1876: 0208142715|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send  7500 of max 31500
	Line 1880: 0208142723|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send 15500 of max 31500
	Line 1884: 0208142739|sip  |3|00|[CTrans::TimeOut100ms] To Server  1 of  1 Retry     INVITE send 31500 of max 31500

You then called

0208142753|sip  |1|00|[CInvite]: szDest  - 8203

 

Which worked.

 

I suggest you take this up with your service provider once you re-visited the FAQ's as the logging levels are not set to the recommended level.

 

If you still struggle please work with above named Reseller to open a ticket.

 

In order to raise a support ticket you need to work with your Polycom reseller as they need to do this for you.

End Customers are unable to open a ticket directly with Polycom support.

As the unit is no longer within warranty please be prepared to Pay Per Incident / PPI. This is all outlined in detail here

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

Please be aware:

The purpose of these forums is to allow community members collaborate and help each other.
Questions posted here do not follow Polycom’s SLA guidelines.
If you require assistance from Polycom technical support, please open a
web service request or call us .

The above is necessary in order to track issue internally within Polycom.

You are welcome to post more questions or configuration or logs for other community members to look at but if your issue requires a fix via Polycom you must go via the official support structure.

Please ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's

Please remember, if you see a post that helped you , and it answers your question, please mark it as an "Accept as Solution".

This forum reply or post is based upon my personal experience and does not reflect the opinion or view of my employer.
Polycom employee participation within this community is not mandatory and any post or FAQ article provided by myself is done either during my working hours or outside working hours, in my private time, and may be answered on weekends, bank holidays or personal holidays.
Message 5 of 10
Occasional Advisor

Re: VVX 600 Unable To Call Out on SIP Line

Thanks @SteffenBaierUK for rescuing me from spam!  I set my Global Log Level Limit to Debug, and my SIP Module Log Level Limit to Level 2.  I then tried to make an 11-digit call (which failed), make a 4-digit call (which succeeded), and receive an inbound call from the same 11-digit number I had previously tried to call (which succeeded).  I copied the portion of the log file related to these calls, which is attached.  I also tried the same calls with the SIP Module Log Level Limit set to Debug, and attached that log excerpt as well.  Anything jump out at you as needing attention?

Highlighted
Polycom Employee & Community Manager

Re: VVX 600 Unable To Call Out on SIP Line

Hello @gurs ,

 

as previsoely outlined I would ask you to take this up with the service provider you are using.

 

0209121515|sip  |0|00|>>> Data Send to 23.253.126.46:5060
0209121515|sip  |0|00|    INVITE sip:13129445852@17683.etollfree-cloud.net:5060;user=phone SIP/2.0
0209121515|sip  |0|00|    Via: SIP/2.0/UDP 192.168.123.163:5060;branch=z9hG4bK56b3be55BD2664C6
0209121515|sip  |0|00|    From: "3123006782" <sip:8201@17683.etollfree-cloud.net>;tag=7C0AF40B-18A4362C
0209121515|sip  |0|00|    To: <sip:13129445852@17683.etollfree-cloud.net;user=phone>
0209121515|sip  |0|00|    CSeq: 1 INVITE
0209121515|sip  |0|00|    Call-ID: a1edc5a9a4311c586b027eaf4cb06cc4
0209121515|sip  |0|00|    Contact: <sip:8201@192.168.123.163:5060>
0209121515|sip  |0|00|    Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
0209121515|sip  |0|00|    User-Agent: PolycomVVX-VVX_600-UA/5.9.1.0615
0209121515|sip  |0|00|    Accept-Language: en
0209121515|sip  |0|00|    Supported: replaces,100rel
0209121515|sip  |0|00|    Allow-Events: conference,talk,hold
0209121515|sip  |0|00|    Max-Forwards: 70
0209121515|sip  |0|00|    Content-Type: application/sdp
0209121515|sip  |0|00|    Content-Length: 534
0209121515|sip  |0|00|    
0209121515|sip  |0|00|    v=0
0209121515|sip  |0|00|    o=- 1549736115 1549736115 IN IP4 192.168.123.163
0209121515|sip  |0|00|    s=Polycom IP Phone
0209121515|sip  |0|00|    c=IN IP4 192.168.123.163
0209121515|sip  |0|00|    b=AS:512
0209121515|sip  |0|00|    t=0 0
0209121515|sip  |0|00|    a=sendrecv
0209121515|sip  |0|00|    m=audio 2222 RTP/AVP 9 102 0 8 18 127
0209121515|sip  |0|00|    a=rtpmap:9 G722/8000
0209121515|sip  |0|00|    a=rtpmap:102 G7221/16000
0209121515|sip  |0|00|    a=fmtp:102 bitrate=32000
0209121515|sip  |0|00|    a=rtpmap:0 PCMU/8000
0209121515|sip  |0|00|    a=rtpmap:8 PCMA/8000
0209121515|sip  |0|00|    a=rtpmap:18 G729/8000
0209121515|sip  |0|00|    a=fmtp:18 annexb=no
0209121515|sip  |0|00|    a=rtpmap:127 telephone-event/8000
0209121515|sip  |0|00|    m=video 2224 RTP/AVP 109 34
0209121515|sip  |0|00|    a=rtpmap:109 H264/90000
0209121515|sip  |0|00|    a=fmtp:109 profile-level-id=42800d; packetization-mode=0
0209121515|sip  |0|00|    a=rtpmap:34 H263/90000
0209121515|sip  |0|00|    a=fmtp:34 CIF=1;QCIF=1;SQCIF=1
0209121515|sip  |0|00|<<< End of data send
0209121515|sip  |2|00|adjustRetransWhenTimerCreated UA Client INVITE INVITE state 'callingTrying' timeout=65 (0x40f02468)
0209121515|sip  |3|00|CStkCall::NewCallState 'Dialtone'->'Proceeding' (0x1bccc38) m_hUI(0x1d7c4e8),Control Channel(0), ResponseCode(-1)
0209121515|sip  |2|00|SipOnEvCallNewState 0x1bccc38,0x1d7c4e8 2,Proceeding, ResponseCode:-1
0209121515|sip  |0|00|listener: Received packet from 23.253.126.46:5060
0209121515|sip  |0|00|listener: Received packet from 23.253.126.46:5060
0209121515|sip  |0|00|<<<Data Received UDP
0209121515|sip  |0|00|    SIP/2.0 100 Trying
0209121515|sip  |0|00|    Via: SIP/2.0/UDP 192.168.123.163:5060;branch=z9hG4bK56b3be55BD2664C6
0209121515|sip  |0|00|    From: "3123006782" <sip:8201@17683.etollfree-cloud.net>;tag=7C0AF40B-18A4362C
0209121515|sip  |0|00|    To: <sip:13129445852@17683.etollfree-cloud.net;user=phone>
0209121515|sip  |0|00|    Call-ID: a1edc5a9a4311c586b027eaf4cb06cc4
0209121515|sip  |0|00|    CSeq: 1 INVITE
0209121515|sip  |0|00|    User-Agent: FreeSWITCH-mod_sofia/1.5.13b+git~20140519T124739Z~ea78f4d0e8~64bit
0209121515|sip  |0|00|    Content-Length: 0
0209121515|sip  |0|00|    
0209121515|sip  |1|00|SipOnCommand: response 100,INVITE fromtag :7C0AF40B-18A4362C toTag :(null)
0209121515|sip  |1|00|SipOnCommand: response 100,INVITE matches user 1 of 1 '8201'
0209121515|sip  |3|00|UA Client INVITE INVITE trans state 'callingTrying'->'proceeding' by 100 resp 65 timeout(0x40f02468)
0209121515|sip  |2|00|[CTrans::ResponseProcess] INVITE InvTran reTrans ALREADY stopped in 'proceeding' state at retryCount 0 code 100, timeout=65 (0x40f02468)
0209121515|sip  |3|00|Use common source preference for incoming and outgoing calls
0209121515|sip  |0|00|[CCommand::NeedToProcessCID] cmdType = 1 cmdMessage = 100 g_csSipRequestSourceMessage = -1 g_csSipResponseSourceMessage = -1---
0209121515|sip  |3|00|GetRemotePartyAddress from 'To'
0209121515|sip  |3|00|CStkCall::OnEvNewDest (0x1bccc38) new display '' user '13129445852' old 'From' new 'To' source
0209121515|sip  |2|00|CStkCall::OnEvSubmitDest CallIdType(1)
0209121515|sip  |0|00|SipOnEvNewDest 0x1bccc38,0x1d7c4e8,13129445852,
0209121515|sip  |3|00|CStkCall::NewCallState 'Proceeding'->'Proceeding' (0x1bccc38) m_hUI(0x1d7c4e8),Control Channel(0), ResponseCode(-1)
0209121515|sip  |2|00|SipOnEvCallNewState 0x1bccc38,0x1d7c4e8 2,Proceeding, ResponseCode:-1
0209121515|clist|4|00|dbCfg::getServerDir:Unknown dbCfg type
0209121515|clist|4|00|dbCfg::getServerDir:Unknown dbCfg type
0209121515|sip  |0|00|<<<Data Received UDP
0209121515|sip  |0|00|    SIP/2.0 407 Proxy Authentication Required
0209121515|sip  |0|00|    Via: SIP/2.0/UDP 192.168.123.163:5060;branch=z9hG4bK56b3be55BD2664C6
0209121515|sip  |0|00|    From: "3123006782" <sip:8201@17683.etollfree-cloud.net>;tag=7C0AF40B-18A4362C
0209121515|sip  |0|00|    To: <sip:13129445852@17683.etollfree-cloud.net;user=phone>;tag=e2Xjc9F8c228p
0209121515|sip  |0|00|    Call-ID: a1edc5a9a4311c586b027eaf4cb06cc4
0209121515|sip  |0|00|    CSeq: 1 INVITE
0209121515|sip  |0|00|    User-Agent: FreeSWITCH-mod_sofia/1.5.13b+git~20140519T124739Z~ea78f4d0e8~64bit
0209121515|sip  |0|00|    Accept: application/sdp
0209121515|sip  |0|00|    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
0209121515|sip  |0|00|    Supported: timer, path, replaces
0209121515|sip  |0|00|    Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
0209121515|sip  |0|00|    Proxy-Authenticate: Digest realm="17683.etollfree-cloud.net", nonce="52489691-af1e-4c7f-ba4a-38ae12a9b625", algorithm=MD5, qop="auth"
0209121515|sip  |0|00|    Content-Length: 0
0209121515|sip  |0|00|    
0209121515|sip  |1|00|SipOnCommand: response 407,INVITE fromtag :7C0AF40B-18A4362C toTag :e2Xjc9F8c228p
0209121515|sip  |1|00|SipOnCommand: response 407,INVITE matches user 1 of 1 '8201'
0209121515|sip  |3|00|UA Client INVITE INVITE trans state 'proceeding'->'completed' by 407 resp 65 timeout(0x40f02468)
0209121515|sip  |3|00|407 challenge received
0209121515|sip  |2|00|SipCallState is not Idle, So send Re-INVITE

0209121515|sip  |2|00|new UA Client INVITE trans state 'callingTrying', timeout=0 (0x40f03868)
0209121515|sip  |1|00|Digest authentication
0209121515|sip  |2|00|CTrans:: SendCommand | ProxyList NOT empty.
0209121515|sip  |2|00|CUser::GetFailBackMode 'Timeout'
0209121515|sip  |1|00|CTrans:: SendCommand | this=0x40f02468, bVQMonMessage=0, m_pCall->m_pUser->m_bOBFailOverReRegOn=0, m_pCall->m_pUser->m_bVQMonFailoverEnabled=1
0209121515|sip  |0|00|Trying to send data to Destination 23.253.126.46 on socket 225
0209121515|sip  |0|00|>>> Data Send to 23.253.126.46:5060
0209121515|sip  |0|00|    ACK sip:13129445852@17683.etollfree-cloud.net:5060;user=phone SIP/2.0
0209121515|sip  |0|00|    Via: SIP/2.0/UDP 192.168.123.163:5060;branch=z9hG4bK56b3be55BD2664C6
0209121515|sip  |0|00|    From: "3123006782" <sip:8201@17683.etollfree-cloud.net>;tag=7C0AF40B-18A4362C
0209121515|sip  |0|00|    To: <sip:13129445852@17683.etollfree-cloud.net;user=phone>;tag=e2Xjc9F8c228p
0209121515|sip  |0|00|    CSeq: 1 ACK
0209121515|sip  |0|00|    Call-ID: a1edc5a9a4311c586b027eaf4cb06cc4
0209121515|sip  |0|00|    Contact: <sip:8201@192.168.123.163:5060>
0209121515|sip  |0|00|    Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
0209121515|sip  |0|00|    User-Agent: PolycomVVX-VVX_600-UA/5.9.1.0615
0209121515|sip  |0|00|    Accept-Language: en
0209121515|sip  |0|00|    Max-Forwards: 70
0209121515|sip  |0|00|    Content-Length: 0
0209121515|sip  |0|00|    
0209121515|sip  |0|00|<<< End of data send
0209121515|sip  |2|00|adjustRetransWhenTimerCreated UA Client INVITE ACK state 'completed' timeout=65 (0x40f02468)
0209121515|sip  |2|00|SendCommand: reqDest '17683.etollfree-cloud.net' isLync 0 isGRUU 0 isIP 0 useEffectiveProxy 1 
0209121515|sip  |1|00|SendCommand: isLync 0 isGRUU 0 isIP 0 useEffectiveProxy 1 
0209121515|sip  |1|00|CreateFailOverProxyList : Reg to Domain '17683.etollfree-cloud.net' nPort 5060, lkup 3
0209121515|sip  |1|00|CreateFailOverProxyList : For INVITE Request nPort 5060
0209121515|sip  |1|00|doDnsListLookup(udp): doDnsSrvLookupForARecordList for '17683.etollfree-cloud.net' port 5060 returned 1 results
0209121515|sip  |1|00|doDnsListLookup(udp): result 0 '23.253.126.46' port 5060 isInBound 0
0209121515|sip  |1|00|CreateFailOverProxyList : 'UDP Only' for '17683.etollfree-cloud.net' port 5060 IP 0 is '23.253.126.46' on udp port 5060
0209121515|sip  |2|00|CUser::GetFailBackMode 'Timeout'
0209121515|sip  |1|00|CreateFailOverProxyList : 'UDP Only' Add rest Total to Try 1
0209121515|sip  |2|00|CreateFailOverProxyList : Exit 'UDP Only' lookup with 1 IP Addresses
0209121515|sip  |2|00|CreateFailOverProxyList : IP 1 is '23.253.126.46' on udp port 5060
0209121515|sip  |2|00|CUser::GetFailBackMode 'Timeout'
0209121515|sip  |1|00|CTrans:: SendCommand | this=0x40f03868, bVQMonMessage=0, m_pCall->m_pUser->m_bOBFailOverReRegOn=0, m_pCall->m_pUser->m_bVQMonFailoverEnabled=1
0209121515|sip  |0|00|Trying to send data to Destination 23.253.126.46 on socket 225
0209121515|sip  |0|00|>>> Data Send to 23.253.126.46:5060
0209121515|sip  |0|00|    INVITE sip:13129445852@17683.etollfree-cloud.net:5060;user=phone SIP/2.0
0209121515|sip  |0|00|    Via: SIP/2.0/UDP 192.168.123.163:5060;branch=z9hG4bKdfed03bfBB468780
0209121515|sip  |0|00|    From: "3123006782" <sip:8201@17683.etollfree-cloud.net>;tag=7C0AF40B-18A4362C
0209121515|sip  |0|00|    To: <sip:13129445852@17683.etollfree-cloud.net;user=phone>
0209121515|sip  |0|00|    CSeq: 2 INVITE
0209121515|sip  |0|00|    Call-ID: a1edc5a9a4311c586b027eaf4cb06cc4
0209121515|sip  |0|00|    Contact: <sip:8201@192.168.123.163:5060>
0209121515|sip  |0|00|    Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
0209121515|sip  |0|00|    User-Agent: PolycomVVX-VVX_600-UA/5.9.1.0615
0209121515|sip  |0|00|    Accept-Language: en
0209121515|sip  |0|00|    Supported: replaces,100rel
0209121515|sip  |0|00|    Allow-Events: conference,talk,hold
0209121515|sip  |0|00|    Proxy-Authorization: Digest username="8201", realm="17683.etollfree-cloud.net", nonce="52489691-af1e-4c7f-ba4a-38ae12a9b625", qop=auth, cnonce="1fym0U6hKHgNyuA", nc=00000001, uri="sip:13129445852@17683.etollfree-cloud.net:5060;user=phone", response="257e387cc678c68a81416de15b5eedae", algorithm=MD5
0209121515|sip  |0|00|    Max-Forwards: 70
0209121515|sip  |0|00|    Content-Type: application/sdp
0209121515|sip  |0|00|    Content-Length: 534
0209121515|sip  |0|00|    
0209121515|sip  |0|00|    v=0
0209121515|sip  |0|00|    o=- 1549736115 1549736115 IN IP4 192.168.123.163
0209121515|sip  |0|00|    s=Polycom IP Phone
0209121515|sip  |0|00|    c=IN IP4 192.168.123.163
0209121515|sip  |0|00|    b=AS:512
0209121515|sip  |0|00|    t=0 0
0209121515|sip  |0|00|    a=sendrecv
0209121515|sip  |0|00|    m=audio 2222 RTP/AVP 9 102 0 8 18 127
0209121515|sip  |0|00|    a=rtpmap:9 G722/8000
0209121515|sip  |0|00|    a=rtpmap:102 G7221/16000
0209121515|sip  |0|00|    a=fmtp:102 bitrate=32000
0209121515|sip  |0|00|    a=rtpmap:0 PCMU/8000
0209121515|sip  |0|00|    a=rtpmap:8 PCMA/8000
0209121515|sip  |0|00|    a=rtpmap:18 G729/8000
0209121515|sip  |0|00|    a=fmtp:18 annexb=no
0209121515|sip  |0|00|    a=rtpmap:127 telephone-event/8000
0209121515|sip  |0|00|    m=video 2224 RTP/AVP 109 34
0209121515|sip  |0|00|    a=rtpmap:109 H264/90000
0209121515|sip  |0|00|    a=fmtp:109 profile-level-id=42800d; packetization-mode=0
0209121515|sip  |0|00|    a=rtpmap:34 H263/90000
0209121515|sip  |0|00|    a=fmtp:34 CIF=1;QCIF=1;SQCIF=1
0209121515|sip  |0|00|<<< End of data send

we get a 407 challenge for our initial INVITE and the Server never responds to our new INVITE

 

You can compare the logs to the working scenario or if all fails follow up as already advised or await any other volunteers to comment.

 

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

Please be aware:

The purpose of these forums is to allow community members collaborate and help each other.
Questions posted here do not follow Polycom’s SLA guidelines.
If you require assistance from Polycom technical support, please open a
web service request or call us .

The above is necessary in order to track issue internally within Polycom.

You are welcome to post more questions or configuration or logs for other community members to look at but if your issue requires a fix via Polycom you must go via the official support structure.

Please ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's

Please remember, if you see a post that helped you , and it answers your question, please mark it as an "Accept as Solution".

This forum reply or post is based upon my personal experience and does not reflect the opinion or view of my employer.
Polycom employee participation within this community is not mandatory and any post or FAQ article provided by myself is done either during my working hours or outside working hours, in my private time, and may be answered on weekends, bank holidays or personal holidays.
Message 7 of 10
Occasional Advisor

Re: VVX 600 Unable To Call Out on SIP Line

Thanks for the feedback @SteffenBaierUK.  As you suggested, I asked the service provider to look into this, but their position is that since two different softphones (X-Lite and UltraSIP) are able to send/receive calls from a PC running on the same network as the Polycom VVX 600, the issue must be somewhere in the Polycom settings.  They are not Polycom experts.  They asked that I do a firmware update and factory-reset on the Polycom, which I have already done, and have confirmed all of the settings I have entered into the Polycom, but otherwise they are not sure how to proceed.

 

If it is helpful, I have attached the debug log from one of the softphones (MicroSIP) while completing the 11-digit outside call referenced above.  Does anything jump out at you?

Polycom Employee & Community Manager

Re: VVX 600 Unable To Call Out on SIP Line

Hello @gurs ,

 

I cannot provide free support and the possible escalation via a PPI has already been outlined.

 

Stating this the only brief difference I can see at present is that the VVX 600 INVITE contains Video and Voice where the Softphone only contains Voice codecs.

 

I suggest you disable video on the phone and test.

 

If this fails the next step has been outlined.

 

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

Please be aware:

The purpose of these forums is to allow community members collaborate and help each other.
Questions posted here do not follow Polycom’s SLA guidelines.
If you require assistance from Polycom technical support, please open a
web service request or call us .

The above is necessary in order to track issue internally within Polycom.

You are welcome to post more questions or configuration or logs for other community members to look at but if your issue requires a fix via Polycom you must go via the official support structure.

Please ensure you always check the VoIP , Video Endpoint , Skype for Business , PSTN or RPM FAQ's

Please remember, if you see a post that helped you , and it answers your question, please mark it as an "Accept as Solution".

This forum reply or post is based upon my personal experience and does not reflect the opinion or view of my employer.
Polycom employee participation within this community is not mandatory and any post or FAQ article provided by myself is done either during my working hours or outside working hours, in my private time, and may be answered on weekends, bank holidays or personal holidays.
Message 9 of 10
Occasional Advisor

Re: VVX 600 Unable To Call Out on SIP Line

EUREKA!!!!!  @SteffenBaierUK, you nailed it!  I disabled Video and Auto Start Video under the Video Processing preferences, rebooted the phone, and outside 11- and 10-digit calling now works like a charm.  That was a great catch.  I have no idea why that would matter given that there is no camera attached to the VVX 600, but I don't really care either.  I am just glad to have the phones working!  Thanks again @SteffenBaierUK.

Message 10 of 10