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- HP Community
- Poly Phones
- Desk and IP Conference Phones
- Directed Call Pickup Only dialing string and not extension

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12-21-2018 03:38 PM
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
Solved! Go to Solution.
Accepted Solutions

12-28-2018 01:49 PM
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
12-22-2018 01:29 AM
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
Notice: I am an HP Poly employee but all replies within the community are done as a volunteer outside of my day role. This community forum is not an official HP Poly support resource, thus responses from HP Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge.
If you need immediate and/or official assistance for former Poly\Plantronics\Polycom please open a service ticket through your support channels
For HP products please check HP Support.
Please also ensure you always check the General VoIP , Video Endpoint , UC Platform (Microsoft) , PSTN

12-28-2018 01:49 PM
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
12-31-2018 05:22 AM
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
Notice: I am an HP Poly employee but all replies within the community are done as a volunteer outside of my day role. This community forum is not an official HP Poly support resource, thus responses from HP Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge.
If you need immediate and/or official assistance for former Poly\Plantronics\Polycom please open a service ticket through your support channels
For HP products please check HP Support.
Please also ensure you always check the General VoIP , Video Endpoint , UC Platform (Microsoft) , PSTN

10-29-2019 08:22 PM
Hi All,
Have been grappling with this issue for some time (had given up in all honesty) but today came across a patch for PJSIP which somebody posted in August this year after trawling through Asterisk Jira.
Have tested it briefly and it seems to work!
Thorough testing has not taken place to confirm it doesn't break other things or cause performance issues, but at least we have directed pickup working in PJSIP.
https://issues.asterisk.org/jira/browse/ASTERISK-24601
Feel free to mark this post as the solution if it works for you.
Cheers,
Tim