Plantronics + Polycom. Now together as Poly Logo

SoundStation IP 600 cannot call SIP-URI

cts
Occasional Contributor

SoundStation IP 600 cannot call SIP-URI

I am trying to place a call to a sip: URI: 

wbdemo@conf.zipdx.com

The same call, from the same network, works fine using a softphone or my built-in router sip-to-dect system.

 

The IP 6000 has the URI in question in the phonebook, I have also tried entering the sip URI directly - no avail. The IP 6000 can call any "real" telephone numbers just fine. The digitmap and digitmap-timout settings are left empty. The phone "tries" the call, stays at the "connecting" phase for around 27 seconds, then gives me the "busy" signal. I have tried entering the sip URI with sip: and without it in my phonebook.xml .

 

What setting have I forgotten in which configuration to be able to place calls to sip: URIs?

 

Thank you in advance,

 Christian.

 

---

Here's the sip debug-level log of the call that didn't go through. I have replaced my sipgate username with 1234567e0 to protect me, the innocent ;)

0131231524|sip  |2|03|SipCallNew 0 local port 2224 call appearance -1 IsRtrv 0 dialog 0
0131231524|sip  |2|03|CStkDialog::CStkDialog SetAddressLocal Config 'cts' <1234567e0@sipgate.de:0>
0131231524|sip  |2|03|CStkDialog::CStkDialog AddressLocal set to Config
0131231524|sip  |3|03|CStkDialog::SetAddressLocal localTag set to ''
0131231524|sip  |3|03|CStkDialog::SetAddressLocal new address added of 1
0131231524|sip  |2|03|CStkDialog::CStkDialog TAG '6BCCC11F-6D46E710' generated
0131231524|sip  |2|03|CStkDialog::CStkDialog local addr 'cts' <1234567e0@sipgate.de:0> Tag '6BCCC11F-6D46E710'
0131231524|sip  |2|03|CStkDialog::CStkDialog exit 0x41496eb4 local list size 1
0131231524|sip  |2|03|CStkDialogList::CreateDialogObject localTarg usr '1234567e0' 
0131231524|sip  |2|03|CUser::CallNew 0x4247b5f0 0x41487fec CallAppr 0 IsRetrieve 0 ThrdParty '' Dialog 0x0 isCentConf 0
0131231524|sip  |3|03|CStkCall::NewCallState reason 16 'Unknown'->'Dialtone' (0x41487fec)
0131231524|sip  |2|03|SipOnEvCallNewState 41487fec,4247b5f0 0,Dialtone
0131231524|sip  |2|03|SipCallMake wbdemo@conf.zipdx.com
0131231524|sip  |2|03|new UA Client INVITE trans state 'callingTrying', timeout=0 (0x41432254)
0131231524|sip  |1|03|CreateFailOverProxyList : Reg to Domain 'sipgate.de' nPort 5060
0131231524|sip  |1|03|CreateFailOverProxyList : For INVITE Request nPort 5060
0131231524|sip  |1|03|doDnsListLookup(udp): doDnsSrvLookupForARecordList for 'sipgate.de' port 5060 returned 1 results
0131231524|sip  |1|03|doDnsListLookup(udp): result 0 '217.10.79.9' port 5060 isInBound 0
0131231524|sip  |1|03|CreateFailOverProxyList : 'UDP Only' for 'sipgate.de' port 5060 IP 0 is '217.10.79.9' on udp port 5060
0131231524|sip  |1|03|CreateFailOverProxyList : 'UDP Only' Add rest Total to Try 1
0131231524|sip  |2|03|CreateFailOverProxyList : Exit 'UDP Only' lookup with 1 IP Addresses
0131231524|sip  |2|03|CreateFailOverProxyList : IP 1 is '217.10.79.9' on udp port 5060
0131231524|sip  |2|03|adjustRetransWhenTimerCreated UA Client INVITE INVITE state 'callingTrying' timeout=65 (0x41432254)
0131231524|sip  |3|03|CStkCall::NewCallState reason 16 'Dialtone'->'Proceeding' (0x41487fec)
0131231524|sip  |2|03|SipOnEvCallNewState 41487fec,4247b5f0 2,Proceeding
0131231524|sip  |2|03|CTrans::InitRetrans for UA Client INVITE INVITE state 'callingTrying' Server 1 of 1 (0x41432254)
0131231524|sip  |2|03|adjustRetransWhenTimerCreated UA Client INVITE INVITE state 'callingTrying' timeout=65 (0x41432254)
0131231525|sip  |3|03|To Server  1 of  1 Retry     INVITE send   400 of max 31400
0131231526|sip  |3|03|To Server  1 of  1 Retry     INVITE send  1400 of max 31400
0131231528|sip  |3|03|To Server  1 of  1 Retry     INVITE send  3400 of max 31400
0131231530|sip  |3|03|Invalid Top Via,From,or To header
0131231532|sip  |3|03|To Server  1 of  1 Retry     INVITE send  7400 of max 31400
0131231540|sip  |3|03|To Server  1 of  1 Retry     INVITE send 15400 of max 31400
0131231544|sip  |3|03|Invalid Top Via,From,or To header
0131231557|sip  |3|03|To Server  1 of  1 Retry     INVITE send 31400 of max 31400
0131231558|sip  |1|03|CTrans::TimeOut500ms m_nMainTimeoutCount == 0. Call SndMsgFail
0131231558|sip  |3|03|CTrans::TimeOut500ms Self Generated 480 Response
0131231558|sip  |3|03|CStateInviteServer::CStateInviteServer central conf user user '' found in contact user '1234567e0' for cent conf URI ''. Set is focus
0131231558|sip  |3|03|CCallBase::OnEvResponse isFocus set for call 0x41487fec
0131231558|sip  |3|03|UA Client INVITE INVITE trans state 'callingTrying'->'completed' by 480 resp 65 timeout(0x41432254)
0131231558|sip  |2|03|CTrans:: INVITE InvTran reTrans ALREADY stopped in 'completed' state at retryCount 323 code 480, timeout=65 (0x41432254)
0131231558|sip  |3|03|CTrans::ResponseProcess Self Generated 480. RROFO Invalidate Registration
0131231558|sip  |3|03|GetRemotePartyAddress from 'To'
0131231558|sip  |3|03|CStkCall::OnEvNewDest (0x41487fec) new display '' user 'wbdemo' old 'From' new 'To' source
0131231558|sip  |2|03|SipOnEvNewDest 41487fec,4247b5f0,sip:wbdemo@conf.zipdx.com,
0131231558|sip  |1|03|Dialog 'id0d6d84b2' State 'Trying'->'Terminated'
0131231558|sip  |3|03|CStkCall::NewCallState reason 14 'Proceeding'->'Idle' (0x41487fec)
0131231558|sip  |2|03|SipOnEvCallNewState 41487fec,4247b5f0 10,service unavailable
0131231558|sip  |1|03|Client State finished INVITE (0x41496eb4)
0131231559|sip  |3|03|Invalid Top Via,From,or To header
0131231600|sip  |2|03|SipCallDrop 41487fec,4247b5f0 reason 6
0131231600|sip  |3|03|CStkCall::Drop(reason = 6) (0x41487fec)
0131231600|sip  |3|03|CStkCall::NewCallState reason 16 'Idle'->'Idle' (0x41487fec)
0131231600|sip  |1|03|CStkCall::NewCallState Already Idle returning (0x41487fec)
0131231600|sip  |1|03|TimeOut500ms Call (0x41487fec) finished
Message 1 of 8
7 REPLIES 7
SteffenBaierUK
Polycom Employee & Community Manager

Re: SoundStation IP 600 cannot call SIP-URI

Hello @cts,

welcome to the Polycom Community.

It is always useful to include the currently used SIP or UC Software version as issues experienced or a question asked may already be addressed in a newer release.

This also allows yourself and others to check against current software release notes, Administrator Guides or FAQ post’s.

The above is also stated in the "Must Read First" and is the absolute minimum requirement every new post should include. .

In addition providing us with this basic information gives Polycom an idea what Software Versions are used in the field and avoids wasting time trying to troubleshoot issues which have already been addressed.

Therefore the Polycom VoIP FAQ contains this post here:

Question: How can I find out my SIP or UC Software Version of my Phone?
Resolution: Please check here

 

The community search finds numerous posts about URL dialing => here <=

 

Can you check if you enabled this?

 

call.urlModeDialing="1"


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

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. All posts and words are my own & do not represent the views of Employer.

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

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 8
cts
Occasional Contributor

Re: SoundStation IP 6000 cannot call SIP-URI

I'm running 4.0.13.1445 on (my typo) a SoundStation 6000 (I forgot the last 0).

 

I have changed my 

0004f2ecxxxx-phone.cfg

 to

 

<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<!-- Application SIP Mink 4.0.13.1445 06-Oct-17 19:45 -->
<!-- Created 12-01-2018 12:00 -->
<PHONE_CONFIG>
<OVERRIDES
np.normal.ringing.calls.tonePattern="ringer14"
call.urlModeDialing="1"
/>
<tone>
<tone.chord>
<tone.chord.callProg>
<tone.chord.callProg.busyTone>
<tone.chord.callProg.busyTone.freq tone.chord.callProg.busyTone.freq.1="425" tone.chord.callProg.busyTone.freq.2="425" />
<tone.chord.callProg.busyTone.level tone.chord.callProg.busyTone.level.1="-20" tone.chord.callProg.busyTone.level.2="-20" />
</tone.chord.callProg.busyTone>
<tone.chord.callProg.dialTone>
<tone.chord.callProg.dialTone.freq tone.chord.callProg.dialTone.freq.1="425" tone.chord.callProg.dialTone.freq.2="425" />
<tone.chord.callProg.dialTone.level tone.chord.callProg.dialTone.level.1="-12" tone.chord.callProg.dialTone.level.2="-12" />
</tone.chord.callProg.dialTone>
<tone.chord.callProg.ringback tone.chord.callProg.ringback.onDur="1000">
<tone.chord.callProg.ringback.freq tone.chord.callProg.ringback.freq.1="425" tone.chord.callProg.ringback.freq.2="425" />
<tone.chord.callProg.ringback.level tone.chord.callProg.ringback.level.1="-20" tone.chord.callProg.ringback.level.2="-20" />
</tone.chord.callProg.ringback>
</tone.chord.callProg>
</tone.chord>
</tone>
</PHONE_CONFIG>

 

I still cannot dial out to SIP URIs, I get a "busy"-signal after a number of seconds.

 

The config file is loaded as I can see from my FTP server.

Message 3 of 8
SteffenBaierUK
Polycom Employee & Community Manager

Re: SoundStation IP 6000 cannot call SIP-URI

Hello @cts,

 

this is my mistake as I provided you with the wrong Parameter.

 

The parameter I meant is:

 

<change feature.urlDialing.enabled="1" />

The above is active in the UC Software 4.0.13 so it will not appear in the configuration when exported.

 

My Test IP7000 is registered to my Asterisk Server but dialing an IP via SpeedDial (10.252.149.60) shows:

 

<135>Feb  6 11:37:15 10.252.149.51 0004f2e693e1|0206113715|so   |1|03|SoMediaSessC::procLclMsg() MsgAppCallDialCmnd sip:10.252.149.60, 0 1 1 4
...
<134>Feb  6 11:37:15 10.252.149.51 0004f2e693e1|0206113715|sip  |2|03|SipCallMake sip:10.252.149.60
...
<135>Feb  6 11:37:15 10.252.149.51 0004f2e693e1|0206113715|sip  |0|03|>>> Data Send to 10.252.149.60:5060
<135>Feb  6 11:37:15 10.252.149.51 0004f2e693e1|0206113715|sip  |0|03|    INVITE sip:10.252.149.60 SIP/2.0
<135>Feb  6 11:37:15 10.252.149.51 0004f2e693e1|0206113715|sip  |0|03|    Via: SIP/2.0/UDP 10.252.149.51;branch=z9hG4bK4945f302F1F05FEF
<135>Feb  6 11:37:15 10.252.149.51 0004f2e693e1|0206113715|sip  |0|03|    From: "3055" <sip:3055@10.252.122.122>;tag=3929F59C-8735BD81
<135>Feb  6 11:37:15 10.252.149.51 0004f2e693e1|0206113715|sip  |0|03|    To: <sip:10.252.149.60>
<135>Feb  6 11:37:15 10.252.149.51 0004f2e693e1|0206113715|sip  |0|03|    CSeq: 1 INVITE

Similar when I dial 123

 

<135>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|so   |1|03|SoMediaSessC::procLclMsg() MsgAppCallDialCmnd 123, 0 1 1 5
...
<135>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|so   |1|03|[SoDigitMapElementC]: Checking 123 (3) against [2-9]11 (3)
<134>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|so   |2|03|[SoDigitMapC]: Disabling [2-9]11 (3) - No match possible (1)
<135>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|so   |1|03|[SoDigitMapElementC]: Checking 123 (3) against 0T (1)
<134>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|so   |2|03|[SoDigitMapC]: Disabling 0T (1) - No match possible (1)
<135>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|so   |1|03|[SoDigitMapElementC]: Checking 123 (3) against 011xxx.T (5)
<134>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|so   |2|03|[SoDigitMapC]: Disabling 011xxx.T (5) - No match possible (1)
<135>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|so   |1|03|[SoDigitMapElementC]: Checking 123 (3) against [0-1][2-9]xxxxxxxxx (11)
<135>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|so   |1|03|[SoDigitMapElementC]: Checking 123 (3) against [2-9]xxxxxxxxx (10)
<134>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|so   |2|03|[SoDigitMapC]: Disabling [2-9]xxxxxxxxx (10) - No match possible (1)
<135>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|so   |1|03|[SoDigitMapElementC]: Checking 123 (3) against [2-9]xxxT (4)
<134>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|so   |2|03|[SoDigitMapC]: Disabling [2-9]xxxT (4) - No match possible (1)
<135>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|so   |1|03|[SoDigitMapElementC]: Checking 123 (3) against **x.T (2)
<134>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|so   |2|03|[SoDigitMapC]: Disabling **x.T (2) - No match possible (1)
...
<134>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|sip  |2|03|SipCallMake 123
...
<135>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|sip  |0|03|>>> Data Send to 10.252.122.122:5060
<135>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|sip  |0|03|    INVITE sip:123@10.252.122.122;user=phone SIP/2.0
<135>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|sip  |0|03|    Via: SIP/2.0/UDP 10.252.149.51;branch=z9hG4bK70f5d616AC45F373
<135>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|sip  |0|03|    From: "3055" <sip:3055@10.252.122.122>;tag=537BDD90-A23BC465
<135>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|sip  |0|03|    To: <sip:123@10.252.122.122;user=phone>
<135>Feb  6 11:37:21 10.252.149.51 0004f2e693e1|0206113721|sip  |0|03|    CSeq: 1 INVITE

We would need to see more logs to help you troubleshoot this.

 

Best Regards

 

Steffen Baier

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. All posts and words are my own & do not represent the views of Employer.

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

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 8
cts
Occasional Contributor

Re: SoundStation IP 6000 cannot call SIP-URI

Hello Steffen,

 

sadly, no avail.

I have added the parameter to my 0004f2xxxxxx-phone.cfg :

<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<!-- Application SIP Mink 4.0.13.1445 06-Oct-17 19:45 -->
<!-- Created 12-01-2018 12:00 -->
<PHONE_CONFIG>
	<OVERRIDES
		np.normal.ringing.calls.tonePattern="ringer14"
		change feature.urlDialing.enabled="1"
	/>
	<tone>
    <tone.chord>
      <tone.chord.callProg>
        <tone.chord.callProg.busyTone>
          <tone.chord.callProg.busyTone.freq tone.chord.callProg.busyTone.freq.1="425" tone.chord.callProg.busyTone.freq.2="425" />
          <tone.chord.callProg.busyTone.level tone.chord.callProg.busyTone.level.1="-20" tone.chord.callProg.busyTone.level.2="-20" />
        </tone.chord.callProg.busyTone>
        <tone.chord.callProg.dialTone>
          <tone.chord.callProg.dialTone.freq tone.chord.callProg.dialTone.freq.1="425" tone.chord.callProg.dialTone.freq.2="425" />
          <tone.chord.callProg.dialTone.level tone.chord.callProg.dialTone.level.1="-12" tone.chord.callProg.dialTone.level.2="-12" />
        </tone.chord.callProg.dialTone>
        <tone.chord.callProg.ringback tone.chord.callProg.ringback.onDur="1000">
          <tone.chord.callProg.ringback.freq tone.chord.callProg.ringback.freq.1="425" tone.chord.callProg.ringback.freq.2="425" />
          <tone.chord.callProg.ringback.level tone.chord.callProg.ringback.level.1="-20" tone.chord.callProg.ringback.level.2="-20" />
        </tone.chord.callProg.ringback>
      </tone.chord.callProg>
    </tone.chord>
  </tone>
</PHONE_CONFIG>

-> nothing.

Both files are indeed loaded from the provisioning ftp server:

[I 2018-02-08 00:57:43] ::ffff:10.3.2.132:32723-[abc] RETR /some/dir/000000000000.cfg completed=1 bytes=1699 seconds=0.025
[I 2018-02-08 00:57:43] ::ffff:10.3.2.132:32723-[abc] RETR /some/dir/0004f2xxxxxx-phone.cfg completed=1 bytes=1474 seconds=0.017
[I 2018-02-08 00:57:43] ::ffff:10.3.2.132:32723-[abc] RETR /some/dir/0004f2xxxxxx-web.cfg completed=1 bytes=967 seconds=0.018

I have then added the parameter to my 0004f2xxxxxx-web.cfg :

<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<!-- Application SIP Mink 4.0.13.1445 06-Oct-17 19:45 -->
<!-- Created 12-01-2018 12:21 -->
<WEB_CONFIG>
	<OVERRIDES
		lcl.ml.lang="German_Germany"
		np.normal.ringing.calls.tonePattern="ringer2"
		pres.idleTimeout.offHours.enabled="0"
		pres.idleTimeout.officeHours.enabled="0"
		up.welcomeSoundEnabled="0"
		voIpProt.SIP.outboundProxy.address="sipgate.de"
		voIpProt.SIP.outboundProxy.port="5060"
		ptt.channel.1.label="PTT 1"
		ptt.channel.24.label="PTT Prio 24"
		ptt.channel.25.label="PTT EMCOM 25"
		ptt.pageMode.group.1.label="Gruppe 1"
		ptt.pageMode.group.24.label="Gruppe Prio"
		ptt.pageMode.group.25.label="Gruppe EMCOM"
		reg.1.address="xxxxxxx"
		reg.1.auth.password="xxxxxx"
		reg.1.auth.userId="xxxxx"
		reg.1.displayName="cts"
		reg.1.label="sipgate"
		voIpProt.server.1.address="sipgate.de"
		voIpProt.server.1.port="5060"
		change feature.urlDialing.enabled="1"
	/>
</WEB_CONFIG>

-> nothing.

Here's the debug level sip trace (looks like before to me):

0208010129|sip  |3|03|Invalid Top Via,From,or To header
0208010131|cfg  |4|03|Prov|Uploading phoneWeb.cfg failed
0208010132|sip  |2|03|SipCallNew 0 local port 2226 call appearance -1 IsRtrv 0 dialog 0
0208010132|sip  |2|03|CStkDialog::CStkDialog SetAddressLocal Config 'cts' <xxxxx@sipgate.de:0>
0208010132|sip  |2|03|CStkDialog::CStkDialog AddressLocal set to Config
0208010132|sip  |3|03|CStkDialog::SetAddressLocal localTag set to ''
0208010132|sip  |3|03|CStkDialog::SetAddressLocal new address added of 1
0208010132|sip  |2|03|CStkDialog::CStkDialog TAG 'BFDBD0BB-74FF2D22' generated
0208010132|sip  |2|03|CStkDialog::CStkDialog local addr 'cts' <xxxxxx@sipgate.de:0> Tag 'BFDBD0BB-74FF2D22'
0208010132|sip  |2|03|CStkDialog::CStkDialog exit 0x41496eb4 local list size 1
0208010132|sip  |2|03|CStkDialogList::CreateDialogObject localTarg usr 'xxxxxxx' 
0208010132|sip  |2|03|CUser::CallNew 0x4247df04 0x41487fec CallAppr 0 IsRetrieve 0 ThrdParty '' Dialog 0x0 isCentConf 0
0208010132|sip  |3|03|CStkCall::NewCallState reason 16 'Unknown'->'Dialtone' (0x41487fec)
0208010132|sip  |2|03|SipOnEvCallNewState 41487fec,4247df04 0,Dialtone
0208010132|sip  |2|03|SipCallMake sip:wbdemo@conf.zipdx.com
0208010132|sip  |2|03|new UA Client INVITE trans state 'callingTrying', timeout=0 (0x41432254)
0208010132|sip  |1|03|CreateFailOverProxyList : Reg to Domain 'sipgate.de' nPort 5060
0208010132|sip  |1|03|CreateFailOverProxyList : For INVITE Request nPort 5060
0208010132|sip  |1|03|doDnsListLookup(udp): doDnsSrvLookupForARecordList for 'sipgate.de' port 5060 returned 1 results
0208010132|sip  |1|03|doDnsListLookup(udp): result 0 '217.10.79.9' port 5060 isInBound 0
0208010132|sip  |1|03|CreateFailOverProxyList : 'UDP Only' for 'sipgate.de' port 5060 IP 0 is '217.10.79.9' on udp port 5060
0208010132|sip  |1|03|CreateFailOverProxyList : 'UDP Only' Add rest Total to Try 1
0208010132|sip  |2|03|CreateFailOverProxyList : Exit 'UDP Only' lookup with 1 IP Addresses
0208010132|sip  |2|03|CreateFailOverProxyList : IP 1 is '217.10.79.9' on udp port 5060
0208010132|sip  |2|03|adjustRetransWhenTimerCreated UA Client INVITE INVITE state 'callingTrying' timeout=65 (0x41432254)
0208010132|sip  |3|03|CStkCall::NewCallState reason 16 'Dialtone'->'Proceeding' (0x41487fec)
0208010132|sip  |2|03|SipOnEvCallNewState 41487fec,4247df04 2,Proceeding
0208010132|sip  |2|03|CTrans::InitRetrans for UA Client INVITE INVITE state 'callingTrying' Server 1 of 1 (0x41432254)
0208010132|sip  |2|03|adjustRetransWhenTimerCreated UA Client INVITE INVITE state 'callingTrying' timeout=65 (0x41432254)
0208010132|sip  |3|03|To Server  1 of  1 Retry     INVITE send   400 of max 31400
0208010133|sip  |3|03|To Server  1 of  1 Retry     INVITE send  1400 of max 31400
0208010135|sip  |3|03|To Server  1 of  1 Retry     INVITE send  3400 of max 31400
0208010139|sip  |3|03|To Server  1 of  1 Retry     INVITE send  7400 of max 31400
0208010144|sip  |3|03|Invalid Top Via,From,or To header
0208010147|sip  |3|03|To Server  1 of  1 Retry     INVITE send 15400 of max 31400
0208010159|sip  |3|03|Invalid Top Via,From,or To header
0208010204|sip  |3|03|To Server  1 of  1 Retry     INVITE send 31400 of max 31400
0208010205|sip  |1|03|CTrans::TimeOut500ms m_nMainTimeoutCount == 0. Call SndMsgFail
0208010205|sip  |3|03|CTrans::TimeOut500ms Self Generated 480 Response
0208010205|sip  |3|03|CStateInviteServer::CStateInviteServer central conf user user '' found in contact user '2550948e0' for cent conf URI ''. Set is focus
0208010205|sip  |3|03|CCallBase::OnEvResponse isFocus set for call 0x41487fec
0208010205|sip  |3|03|UA Client INVITE INVITE trans state 'callingTrying'->'completed' by 480 resp 65 timeout(0x41432254)
0208010205|sip  |2|03|CTrans:: INVITE InvTran reTrans ALREADY stopped in 'completed' state at retryCount 321 code 480, timeout=65 (0x41432254)
0208010205|sip  |3|03|CTrans::ResponseProcess Self Generated 480. RROFO Invalidate Registration
0208010205|sip  |3|03|GetRemotePartyAddress from 'To'
0208010205|sip  |3|03|CStkCall::OnEvNewDest (0x41487fec) new display '' user 'wbdemo' old 'From' new 'To' source
0208010205|sip  |2|03|SipOnEvNewDest 41487fec,4247df04,sip:wbdemo@conf.zipdx.com,
0208010205|sip  |1|03|Dialog 'ida2d5bfc8' State 'Trying'->'Terminated'
0208010205|sip  |3|03|CStkCall::NewCallState reason 14 'Proceeding'->'Idle' (0x41487fec)
0208010205|sip  |2|03|SipOnEvCallNewState 41487fec,4247df04 10,service unavailable
0208010205|sip  |1|03|Client State finished INVITE (0x41496eb4)
0208010207|sip  |2|03|SipCallDrop 41487fec,4247df04 reason 6
0208010207|sip  |3|03|CStkCall::Drop(reason = 6) (0x41487fec)
0208010207|sip  |3|03|CStkCall::NewCallState reason 16 'Idle'->'Idle' (0x41487fec)
0208010207|sip  |1|03|CStkCall::NewCallState Already Idle returning (0x41487fec)
0208010208|sip  |1|03|TimeOut500ms Call (0x41487fec) finished

Is there anything you see from the debug trace that's still going wrong? The behavior is as before: I dial the SIP URI, the nothing for a number of seconds, the a busy signal and I need to "hang up".

 

Thank you,

 Christian.

Message 5 of 8
SteffenBaierUK
Polycom Employee & Community Manager

Re: SoundStation IP 6000 cannot call SIP-URI

Hello @cts,

Again the parameter is already set to the default value of 1 so adding it (you are using the wrong context anyway) will not change anything as it is already active.

 

Looking at your logs (not in debug) I see:

 

0208010205|sip  |3|03|CTrans::TimeOut500ms Self Generated 480 Response

 

We would need to see debug logs for this or span a port and take a wireshark trace.

Best Regards

Steffen Baier

Polycom Global Services

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. All posts and words are my own & do not represent the views of Employer.

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

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 6 of 8
cts
Occasional Contributor

Re: SoundStation IP 6000 cannot call SIP-URI

As for the sip part, I am already on the highest debug level the phone allows for.

Would you like any other component to be logged, and if yes, at which verbosity level?

I'd always try not do disclose a full wireshark trace to a public forum - is there a way I can send you the trace non-publicly?

Also, to better my understanding of the 'contexts' in which a parameter has to be in the three config files that I know of - how would a correct config file as per my above settings look like?

 

Thanks,

 Christian.

Message 7 of 8
SteffenBaierUK
Polycom Employee & Community Manager

Re: SoundStation IP 6000 cannot call SIP-URI

Hello @cts,

The community's VoIP FAQ contains this post here:

 

Oct 11, 2011 Question: How can I troubleshoot configuration Issues via Log Files or the phone GUI?

Resolution: Please check => here <=

 

and

 

Oct 17, 2011 Question: How can change Logging Levels or use Syslog?

Resolution: Please check => here <=

 

Both of the above can be used to help end users to troubleshoot issues.

 

You can attach log files or wireshark traces but you must not assume that Polycom Employees look at these or help you to fix your issue.

 

Mar 8, 2012 Question: What kind of support should I expect from the Community?
Clarification: Please check => here <=

 

The above explains this in detail.

 

For your configuration parameter:

 

 

change feature.urlDialing.enabled="1"

The Word change does not exist in our configuration parameters and neither in XML. You cannot use this.

 

 

The actual Parameter feature.urlDialing.enabled="1" is the default setting within the software so adding this will not do anything.

 

A backup of the phone will not contain this as this is already set to ="1" in the phone software itself.

 

In addition you should never make any changes to the <mac>-web.cfg or <mac>-phone.cfg files as these are uploaded by the phone.

 

Mar 08, 2013 QuestionWhat files does my phone download or upload and why?

Resolution: Please check => here <=

 

The above FAQ has details on this.

 

I could not find any details on the MAC 0004f2e693e1 so I am unable to tell you your reseller.

 

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.

Best Regards

Steffen Baier

Polycom Global Services

----------------
The title Polycom Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. All posts and words are my own & do not represent the views of Employer.

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

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 8 of 8