Directed Call Pickup Only dialing string and not extension

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Occasional Visitor

Directed Call Pickup Only dialing string and not extension

We are trying to set up a new system with all vvx series phones  410/500/450  however i keep running into a very large problem for us.  I have set up BLF keys to monitor other extensions and need to be able to perform directed call pickup using the blf key.  We are able to perform the directed call pickup using the manual key sequence  **ext#.  However when using the softkey the phone only dials the entered string from the config **  and nothing else.  The phone remains at the dial screen with ** entered. I have set the directed call pickup paramters as legacy, and the correct string, Is there somethign I am missing?    I tried attaching Config file however it won't allow text files and software as follows

UC Software Version 5.8.2.4732 Updater Version 5.9.6.10658

 

Thanks

 

-Jace

Message 1 of 4
3 REPLIES
Polycom Employee & Community Manager

Re: Directed Call Pickup Only dialing string and not extension

Hello @jataylor,

 

welcome to the Polycom Community.

Both the community's Must Read First and the community's 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,Skype for Business Online, Skype for Business on Premise, Lync)
  • Additional Polycom 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

  • 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 if the issue is a day 1 issue or only happened after an upgrade or any other relevant details

Whilst providing some of these details may not directly impact any possible answer the community can provide, it does enable Polycom 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 a potential support partners if an issue needs to come into support. It also enables us to verify the entitlement for using features.


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

 

Nothing attached.

 

Nov 4, 2011 Question: How can I setup a Busy Lamp Field / BLF with an Digium Asterisk SIP Server?

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

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 4
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Occasional Visitor

Re: Directed Call Pickup Only dialing string and not extension

Thank you for the help with your link...  I bellieve it's worth mentioning and adding to the page that you've linked to, that It is NOT possible to do directed call pickup with the polycom phones using PJSIP.  This was my actual problem.  I had notifycid=yes in my config, however,  that parameter only applies to chan_sip  not PJSIP.  Again in order to get directed call pickup to work you MUST use CHAN_SIP  not PJSIP for your extensions.

 

Thanks again

 

-Jace

Message 3 of 4
Polycom Employee & Community Manager

Re: Directed Call Pickup Only dialing string and not extension

Hello @jataylor,

 

we had a similar discussion using PJSIP => here <=

 

Looking at the NOTIFY created by PJSIP:

 

1231104332|sip  |0|00|    NOTIFY sip:3031@10.252.149.56 SIP/2.0
1231104332|sip  |0|00|    Via: SIP/2.0/UDP 10.252.149.247:5060;rport;branch=z9hG4bKPjc99744a1-5ec3-4d37-ad7b-7e863f6b7163
1231104332|sip  |0|00|    From: <sip:3030@10.252.149.247>;tag=a36d0efe-f52d-46cd-aa7e-6161b6442bba
1231104332|sip  |0|00|    To: "3031" <sip:3031@10.252.149.247>;tag=995F5897-7A285DA8
1231104332|sip  |0|00|    Contact: <sip:10.252.149.247:5060>
1231104332|sip  |0|00|    Call-ID: b72a65ac2245d3514b3caac9a662be84
1231104332|sip  |0|00|    CSeq: 7685 NOTIFY
1231104332|sip  |0|00|    Event: dialog
1231104332|sip  |0|00|    Subscription-State: active;expires=1900
1231104332|sip  |0|00|    Allow-Events: presence, dialog, message-summary, refer
1231104332|sip  |0|00|    Max-Forwards: 70
1231104332|sip  |0|00|    User-Agent: Asterisk PBX 13.11.2
1231104332|sip  |0|00|    Content-Type: application/dialog-info+xml
1231104332|sip  |0|00|    Content-Length:   250
1231104332|sip  |0|00|    
1231104332|sip  |0|00|    <?xml version="1.0" encoding="UTF-8"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="3" state="full" entity="sip:3030@10.252.149.247:5060">
 <dialog id="3030" direction="recipient">
  <state>early</state>
 </dialog>
</dialog-info>

I only had a brief look at this and it seems you may be able to make changes to the pjsip.conf in relation to:

 

send_pai=yes  ; Send the P Asserted Identity header (default: "no")
send_rpid=yes   ; Send the Remote Party ID header (default: "no")


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 4 of 4