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- IP550 BLF call pickup

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08-02-2017 04:50 PM
feature.directedCallPickup.enabled="1" voIpProt.SIP.useLocalTargetUriForLegacyPickup="1" call.directedCallPickupMethod="legacy" call.directedCallPickupString="*8" attendant.resourceList.1.address="103"
attendant.resourceList.1.type="automata"
also tried normal
My BLF does not send the digits after the call pickup string, it only sends *8. There is nothing in the packet that could possibly indicate that the phone wants to pickup the call coming in on 103. This should be possible according to the FAQ http://community.polycom.com/t5/VoIP/FAQ-Busy-Lamp-Field-for-SoundPoint-IP-supported-Phones-on-a/td-...
It does this even from the factory reset state on the latest firmware.
Solved! Go to Solution.
Accepted Solutions

08-04-2017 03:54 PM
Actually, I have found some old Asterisk source code for SIP which suggests I need both of those XML bits.
http://doxygen.asterisk.org/trunk/d5/dfe/chan__sip_8c.html line 14728.
I will see if I can make my SIP proxy add this XML to the SIP body when the user agent is Polycom.
I still think it would be better to do what every other manufacturer does and prefix the value of the BLF with the pickup string and send that rather than requesting information in a notify, maybe it does on some of the newer models I don't know. We don't actually use Polycom devices or plan to, but I though it would be nice to support them anyway.
Thanks for the help.
08-02-2017 11:53 PM
Hello fortytwo,
Stating the latest software does not help others when finding this post in days, weeks, months or years to come as they do not know what software mentioned was current at the time of writing.
Please therefore always post the exact complete build id.
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
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

08-03-2017 08:26 AM
I've got a Soundpoint IP550 on 4.0.1.13681 and a VVX300 on 4.1.4.7430.
Here is the INVITE sent by the BLF when trying to pickup a call from extension 100 to extension 103, I would expect that the To user should be *8103.
Via: SIP/2.0/UDP 192.168.1.183;branch=z9hG4bKf958097574CB8026 From: "106" <sip:106@server.net>;tag=925F4C17-5D1A4220 To: <sip:*8@server.net;user=phone> CSeq: 1 INVITE Call-ID: a77dcb03-39184e9c-5ed13441@192.168.1.183 Contact: <sip:106@192.168.1.183> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/4.0.1.13681 Accept-Language: en Supported: 100rel,replaces Allow-Events: conference,talk,hold Max-Forwards: 70 Content-Type: application/sdp Content-Length: 344 v=0 o=- 1501769825 1501769825 IN IP4 192.168.1.183 s=Polycom IP Phone c=IN IP4 192.168.1.183 t=0 0 a=sendrecv m=audio 2236 RTP/AVP 0 110 18 9 8 127 a=rtpmap:0 PCMU/8000 a=rtpmap:110 iLBC/8000 a=fmtp:110 mode=20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:127 telephone-event/8000
08-03-2017 08:31 AM
Hello fortytwo,
Your original post stated you are using the latest software which would be 4.0.12 but above explains 4.0.1.XXXX instead which is not the latest software for the SPIP550. In addition UC Software 5.6.0 is the currently latest for the VVX range.
I added some additional details to the Asterisk BLF FAQ post as the NOTIFY send by your yet unknown Server needs to add additional information in order for this to work:
The below notifycid=yes causes Asterisk to add a "target uri = callID" to the XML of the SIP NOTIFY. Without this target uri the Polycom phone will not do a directed pickup.
Any other compatible SIP server also needs to utilize this functionality within their SIP NOTIFY.
Does your NOTIFY contain this ?
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

08-04-2017 03:44 PM
We are using PJSIP for Asterisk which does not have notifycid, I suppose I could try adding XML to the notify in our SIP prxy but that is a very hacky way of doing things.
It would be nice if Polycom was capable of using the value of the BLF as the target, it is already there for initiating calls from the BLF...
Is it the dialog information that is needed? callID, to-tag, from-tag
<dialog id="3081" call-id="callid@ip" local-tag="to-tag" remote-tag="from-tag" direction="recipient">
Or should I be needing the remote or local target URI?
<remote> <identity display="Spectralink 8030 00907a07c46c">sip:3001@10.252.122.122</identity> <target uri="sip:3001@10.252.122.122"/> </remote> <local> <identity>sip:3081@10.252.122.122</identity> <target uri="sip:3081@10.252.122.122"/> </local>
All of this data is already contained in the SIP header so it seems a little silly that it would be needed in the SIP body again...
I guess that might make it easy for my SIP proxy to add them in.

08-04-2017 03:54 PM
Actually, I have found some old Asterisk source code for SIP which suggests I need both of those XML bits.
http://doxygen.asterisk.org/trunk/d5/dfe/chan__sip_8c.html line 14728.
I will see if I can make my SIP proxy add this XML to the SIP body when the user agent is Polycom.
I still think it would be better to do what every other manufacturer does and prefix the value of the BLF with the pickup string and send that rather than requesting information in a notify, maybe it does on some of the newer models I don't know. We don't actually use Polycom devices or plan to, but I though it would be nice to support them anyway.
Thanks for the help.
08-05-2017 02:12 AM
Hello fortytwo,
any standard Asterisk Server has this feature and we have been supporting this feature aka SUBSCRIBE and NOTIFY as an RFC forever.
If you believe we should also add your method and you have a valid business plan please follow this:
Jan 03, 2013 Question: How can I request a change to the current Polycom SIP / UC Software?
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