BLF Park Retreive

[SamFiller]

Let me preface this with this is not a general issue as I have 1000's of phones without an issue.

I have a particular set of phones that are having trouble retrieving a parked call via one touch retrieve.

I tested on multiple firmwares currently testing on a VVX401 6.0.0.4839

A call comes in, I answer it, I push the park button - The call gets parked.

I push the "red" park button to retreive and I get the "dial dialog" (Enter numer with recent outbound calls)

If I hold down the red button it says

Remote: Park 2

Soft button 1: Retreive  Soft button 2: Dial

If I hit Retreive it does the same thing (above)

If I hit Dial it calls the park and it retreived the call.

The question is what logging can I set to see why the standard retreive is not wokring?

Currently the only thing i see in the logs

1104194930|sip  |3|00|CCallNoCall::NewCallState 'ServicePing'->'Idle' (0x15c2b58)
1104194930|sip  |3|00|CStateOptionClient::OnEvResponse , received response code 407
1104194930|sip  |1|00|CStkDialog::SetDialogState: Dialog 'id00c82cd6' State 'Trying'->'Terminated'
1104194930|so   |4|00|MsgAppPickupRemoteCall: Current Media Session is not in Active List.
1104194930|app1 |4|00|[AppCallC::Call] Outgoing Call is not allowed. pAppUi->getBtoeControlChannelStatus() [0]
1104194930|sip  |2|00|SipCallNew 0 local port 2242 video port 0 SDP (nil) call appearance -1 IsRtrv 0 dialog (nil) isDectCall 0 pBargeInfo (nil)
1104194930|sip  |2|00|SipCallNew intercom enabled:0
1104194930|sip  |3|00|User[0x40f24cc8] IsSARenewInProgress[NO]

Any help appriciated

[SteffenBaierUK]

Hello @SamFiller ,

Welcome to the Poly Community.

Both the communities Must Read First and the FAQ reference the basic minimum information a new or follow up post should contain.

This ensures the questions having to be asked are limited and any new or follow up post contains the right amount of details to ensure any voluntary participant within the community does not spend additional time chasing basic information.

As a reminder the basic information asked for:

• Provide the exact Software Version of your Phone
• Provide the Phone Model
• Provide the Call Platform (aka openSIP,Teams, Skype for Business Online, Skype for Business on Premise, Lync, Zoom or BlueJeans)
• Provide details if UC / SIP or ObiEdition
• Additional Poly Infrastructure (RPRM,PDMS or BToE)
• If applicable provide a backup of the phone in question

UC Software 4.0.0 or later via the Web Interface Utilities > Phone Backup & Restore > Phone Backup > Phone Backup. Please rename into .TXT or Zip the file to attach.
Since UC Software 5.9.0 simply provide this via the Web Interface Diagnostics > Download Support Information Package

• If possible provide a Log and either attach them or use the Code Tag.Consult the Troubleshooting Section found within the FAQ if applicable
• If possible provide the MAC Address or Serial of the device
• Provide details for example if the issue is a day 1 issue or only happened after an upgrade or any other relevant details
• For questions around Support please check here

Whilst providing some of these details may not directly impact any possible answer the community can provide, it does enable Poly to have an overview of the current software used. In addition, providing all details at the same time allow us to check logs or look up potential support partners if an issue needs to come into support. It also enables us to verify the entitlement for using features.

I can see that you are a service provider so usually, I would suspect that you can work directly with our support team on something like this.

Your post sounds like we should know what red button you are talking about but I assume we do not. Not having your configuration either makes things difficult for the volunteers within this community to grasp what you are talking about.

Please ensure you always check the FAQ's and/or utilize the community search before posting any new topics or follow up posts.

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

Resolution: Please check => here <=

or

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

Resolution: Please check => 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

[SamFiller]

@SteffenBaierUK

Thank you- as mentioned VVX401  6.0.0.4839 

The softswitch is proprietary but more like OpenSIP

I am familiar with logging :-) and the already set the SIP to DEBUG

My questions was more what other logging should I enable specifically for this issue as SIP is not showing me any issues.

The issue again is specific to "reteive" why is it showing an "Enter number" screen as appose to retreiving.

What is the phone doing when trying to "retreive " that its "erroring" and not doing anything.

[SteffenBaierUK]

Hello @SamFiller ,

as you have not yet attached a backup of your phone we cannot verify the logging settings. We are therefore unable to comment why your logs do not show the SIP logging at debug level.

The next step would be to open a ticket.

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

If this is some sort of an Internet discounter providing your MAC address or your Poly devices serial will enable us to look up who would be able to support you. This may not be who you purchased the Poly device from.

If the unit is no longer within the 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

[SamFiller]

Thank you-

I am gettting SIP in the debug - its this particular issue is not showing and that may be beacuse its not a "SIP" issue.

I can certanly open a ticket, but was thinking if this was here it may help others.

[SamFiller]

This may be the log line I am looking for, but unfortantly unsure how to translate

1105171645|sip  |1|00|CStkDialog::TimeOut500ms: Client State txn finished REGISTER (0x15de548), z9hG4bK7635bd5030530B27
1105171645|sip  |1|00|CStkDialog::TimeOut500ms: Client State txn finished REGISTER (0x15de548), z9hG4bK7306437279E79269
1105171649|so   |4|00|MsgAppPickupRemoteCall: Current Media Session is not in Active List.
1105171649|app1 |4|00|[AppCallC::Call] Outgoing Call is not allowed. pAppUi->getBtoeControlChannelStatus() [0]
1105171650|sip  |2|00|SipCallNew 0 local port 2248 video port 0 SDP (nil) call appearance -1 IsRtrv 0 dialog (nil) isDectCall 0 pBargeInfo (nil)
1105171650|sip  |2|00|SipCallNew intercom enabled:0
1105171650|sip  |3|00|User[0x40f0d708] IsSARenewInProgress[NO]

[SteffenBaierUK]

Hello @SamFiller ,

again this log snipped does not show SIP at debug. It shows SIP at Event 1.

Please work this with our support team as I am personally unable to provide support via the community.

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

Best Regards

Steffen Baier

[SamFiller]

Ok - I didnt get a chance to open a ticket, and hope to open one up so its fixed in future firmware versions.

In short the issue is in the original notify to the phone.

The "confirmed" notify regarding the BLF has an "&" in the dispaly name.

The phone doesnt like that and will not "retreive" the call.

Message Body
    <?xml version="1.0"?>\n
    <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="18" state="full" entity="sip:9705@domain.com"\n
    >\n
    <dialog id="201911052243045555-298b6f9f8d8b969789765400c6a8f748">\n
    <state>confirmed</state>\n
    <local>\n
    <identity display="First & ASSOC">"sip:9705@domain.com"\n
    </identity>\n
    <target uri="sip:9705@domain.com"/>\n
    </local>\n
    <remote>\n
    <identity>"sip:100@domain2.com"\n
    </identity>\n
    <target uri="sip:100@domain2.com"/>\n
    </remote>\n
    </dialog>\n
    </dialog-info>\n 

[SamFiller]

I have filed this as a BUG on my end, as my switch should be sending BLF Dialog in XML encoding

<identity display="First &amp; ASSOC">"sip:9705@domain.com"\n
Vs
<identity display="First & ASSOC">"sip:9705@domain.com"\n

You can see here: https://tools.ietf.org/html/rfc4235#page-17  - on page 16