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Good day. Please help me as I am frustruated, hopeless and clueless. I have a setup with several IP phones and asterisk pbx in a local network. Everything worked for years and then suddenly stopped working - i can place a call from any phone to outside world, but no phone receives a call. I do not know what happened - I blame the asterisk update but I have no clue what's going on.

For simplicity let's say I only have two phones. Phone with extension "10", ip address 192.168.2.30 and phone with extension "11", ip address 192.168.2.31 . Asterisk server is 192.168.2.1 . All local, no NAT involved.

 

Both phones register just fine, here is a registration log for the first phone, second is identical:

Frame 6: 624 bytes on wire (4992 bits), 624 bytes captured (4992 bits)
Ethernet II, Src: Polycom_68:55:6d (00:04:f2:68:55:6d), Dst: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3)
Internet Protocol Version 4, Src: 192.168.2.30, Dst: 192.168.2.1
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol (REGISTER)
    Request-Line: REGISTER sip:192.168.2.1:5060 SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 192.168.2.30;branch=z9hG4bKe2446e3941C6E164
        From: "10" <sip:10@192.168.2.1>;tag=AF63E766-976C5899
        To: <sip:10@192.168.2.1>
        CSeq: 33 REGISTER
        Call-ID: 37a3fb02-9dc0b095-d978f620@192.168.2.30
        Contact: <sip:10@192.168.2.30>;expires=0
        User-Agent: PolycomSoundPointIP-SPIP_335-UA/4.1.1.0731
        Accept-Language: en
        Authorization: Digest username="10", realm="asterisk", nonce="6981c9ac", uri="sip:192.168.2.1:5060", response="b0161e3673e6f6d6a0b51a39b0800791", algorithm=MD5
        Max-Forwards: 70
        Expires: 0
        Content-Length: 0
Frame 7: 589 bytes on wire (4712 bits), 589 bytes captured (4712 bits)
Ethernet II, Src: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3), Dst: Polycom_68:55:6d (00:04:f2:68:55:6d)
Internet Protocol Version 4, Src: 192.168.2.1, Dst: 192.168.2.30
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol (401)
    Status-Line: SIP/2.0 401 Unauthorized
    Message Header
        Via: SIP/2.0/UDP 192.168.2.30;branch=z9hG4bKe2446e3941C6E164;received=192.168.2.30;rport=5060
        From: "10" <sip:10@192.168.2.1>;tag=AF63E766-976C5899
        To: <sip:10@192.168.2.1>;tag=as4a5549d0
        Call-ID: 37a3fb02-9dc0b095-d978f620@192.168.2.30
        CSeq: 33 REGISTER
        Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
        Supported: replaces, timer
        WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35a41ac9"
        Content-Length: 0
Frame 12: 624 bytes on wire (4992 bits), 624 bytes captured (4992 bits)
Ethernet II, Src: Polycom_68:55:6d (00:04:f2:68:55:6d), Dst: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3)
Internet Protocol Version 4, Src: 192.168.2.30, Dst: 192.168.2.1
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol (REGISTER)
    Request-Line: REGISTER sip:192.168.2.1:5060 SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 192.168.2.30;branch=z9hG4bKb51f0c474E598BA2
        From: "10" <sip:10@192.168.2.1>;tag=AF63E766-976C5899
        To: <sip:10@192.168.2.1>
        CSeq: 34 REGISTER
        Call-ID: 37a3fb02-9dc0b095-d978f620@192.168.2.30
        Contact: <sip:10@192.168.2.30>;expires=0
        User-Agent: PolycomSoundPointIP-SPIP_335-UA/4.1.1.0731
        Accept-Language: en
        Authorization: Digest username="10", realm="asterisk", nonce="35a41ac9", uri="sip:192.168.2.1:5060", response="wonttellyousorry", algorithm=MD5
        Max-Forwards: 70
        Expires: 0
        Content-Length: 0
Frame 66: 552 bytes on wire (4416 bits), 552 bytes captured (4416 bits)
Ethernet II, Src: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3), Dst: Polycom_68:55:6d (00:04:f2:68:55:6d)
Internet Protocol Version 4, Src: 192.168.2.1, Dst: 192.168.2.30
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol (200)
    Status-Line: SIP/2.0 200 OK
    Message Header
        Via: SIP/2.0/UDP 192.168.2.30;branch=z9hG4bKb51f0c474E598BA2;received=192.168.2.30;rport=5060
        From: "10" <sip:10@192.168.2.1>;tag=AF63E766-976C5899
        To: <sip:10@192.168.2.1>;tag=as4a5549d0
        Call-ID: 37a3fb02-9dc0b095-d978f620@192.168.2.30
        CSeq: 34 REGISTER
        Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
        Supported: replaces, timer
        Expires: 0
        Date: Sun, 22 Jan 2017 07:07:56 GMT
        Content-Length: 0

When I make a call from one phone to another, it seems that the called phone just ignores all packets from asterisk. So, calling extension 11 from 10 produces the expected handshake:

Frame 1: 949 bytes on wire (7592 bits), 949 bytes captured (7592 bits)
Ethernet II, Src: Polycom_68:55:6d (00:04:f2:68:55:6d), Dst: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3)
Internet Protocol Version 4, Src: 192.168.2.30, Dst: 192.168.2.1
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:11@192.168.2.1:5060;user=phone SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 192.168.2.30;branch=z9hG4bKe257836c8114578F
        From: "10" <sip:10@192.168.2.1>;tag=D9B2C98E-59EF4901
        To: <sip:11@192.168.2.1;user=phone>
        CSeq: 1 INVITE
        Call-ID: 50fbb22a-8f4bdfd-3f1393c8@192.168.2.30
        Contact: <sip:10@192.168.2.30>
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
        User-Agent: PolycomSoundPointIP-SPIP_335-UA/4.1.1.0731
        Accept-Language: en
        Supported: 100rel,replaces
        Allow-Events: conference,talk,hold
        Max-Forwards: 70
        Content-Type: application/sdp
        Content-Length: 294
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): - 1485014073 1485014073 IN IP4 192.168.2.30
            Session Name (s): Polycom IP Phone
            Connection Information (c): IN IP4 192.168.2.30
            Time Description, active time (t): 0 0
            Session Attribute (a): sendrecv
            Media Description, name and address (m): audio 2224 RTP/AVP 9 0 8 18 127
            Media Attribute (a): rtpmap:9 G722/8000
            Media Attribute (a): rtpmap:0 PCMU/8000
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): rtpmap:18 G729/8000
            Media Attribute (a): fmtp:18 annexb=no
            Media Attribute (a): rtpmap:127 telephone-event/8000
Frame 2: 42 bytes on wire (336 bits), 42 bytes captured (336 bits)
Ethernet II, Src: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3), Dst: Broadcast (ff:ff:ff:ff:ff:ff)
Address Resolution Protocol (request)
    Hardware type: Ethernet (1)
    Protocol type: IPv4 (0x0800)
    Hardware size: 6
    Protocol size: 4
    Opcode: request (1)
    Sender MAC address: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3)
    Sender IP address: 192.168.2.1
    Target MAC address: 00:00:00_00:00:00 (00:00:00:00:00:00)
    Target IP address: 192.168.2.30
Frame 3: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
Ethernet II, Src: Polycom_68:55:6d (00:04:f2:68:55:6d), Dst: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3)
Address Resolution Protocol (reply)
    Hardware type: Ethernet (1)
    Protocol type: IPv4 (0x0800)
    Hardware size: 6
    Protocol size: 4
    Opcode: reply (2)
    Sender MAC address: Polycom_68:55:6d (00:04:f2:68:55:6d)
    Sender IP address: 192.168.2.30
    Target MAC address: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3)
    Target IP address: 192.168.2.1
Frame 4: 596 bytes on wire (4768 bits), 596 bytes captured (4768 bits)
Ethernet II, Src: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3), Dst: Polycom_68:55:6d (00:04:f2:68:55:6d)
Internet Protocol Version 4, Src: 192.168.2.1, Dst: 192.168.2.30
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol (401)
    Status-Line: SIP/2.0 401 Unauthorized
    Message Header
        Via: SIP/2.0/UDP 192.168.2.30;branch=z9hG4bKe257836c8114578F;received=192.168.2.30;rport=5060
        From: "10" <sip:10@192.168.2.1>;tag=D9B2C98E-59EF4901
        To: <sip:11@192.168.2.1;user=phone>;tag=as7cca87a4
        Call-ID: 50fbb22a-8f4bdfd-3f1393c8@192.168.2.30
        CSeq: 1 INVITE
        Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
        Supported: replaces, timer
        WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4ebb265d"
        Content-Length: 0
Frame 5: 567 bytes on wire (4536 bits), 567 bytes captured (4536 bits)
Ethernet II, Src: Polycom_68:55:6d (00:04:f2:68:55:6d), Dst: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3)
Internet Protocol Version 4, Src: 192.168.2.30, Dst: 192.168.2.1
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol (ACK)
    Request-Line: ACK sip:11@192.168.2.1:5060;user=phone SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 192.168.2.30;branch=z9hG4bKe257836c8114578F
        From: "10" <sip:10@192.168.2.1>;tag=D9B2C98E-59EF4901
        To: <sip:11@192.168.2.1;user=phone>;tag=as7cca87a4
        CSeq: 1 ACK
        Call-ID: 50fbb22a-8f4bdfd-3f1393c8@192.168.2.30
        Contact: <sip:10@192.168.2.30>
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
        User-Agent: PolycomSoundPointIP-SPIP_335-UA/4.1.1.0731
        Accept-Language: en
        Max-Forwards: 70
        Content-Length: 0
Frame 6: 1123 bytes on wire (8984 bits), 1123 bytes captured (8984 bits)
Ethernet II, Src: Polycom_68:55:6d (00:04:f2:68:55:6d), Dst: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3)
Internet Protocol Version 4, Src: 192.168.2.30, Dst: 192.168.2.1
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:11@192.168.2.1:5060;user=phone SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 192.168.2.30;branch=z9hG4bK6986a4b12DE4646
        From: "10" <sip:10@192.168.2.1>;tag=D9B2C98E-59EF4901
        To: <sip:11@192.168.2.1;user=phone>
        CSeq: 2 INVITE
        Call-ID: 50fbb22a-8f4bdfd-3f1393c8@192.168.2.30
        Contact: <sip:10@192.168.2.30>
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
        User-Agent: PolycomSoundPointIP-SPIP_335-UA/4.1.1.0731
        Accept-Language: en
        Supported: 100rel,replaces
        Allow-Events: conference,talk,hold
        Authorization: Digest username="10", realm="asterisk", nonce="4ebb265d", uri="sip:11@192.168.2.1:5060;user=phone", response="nopeagain", algorithm=MD5
        Max-Forwards: 70
        Content-Type: application/sdp
        Content-Length: 294
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): - 1485014073 1485014073 IN IP4 192.168.2.30
            Session Name (s): Polycom IP Phone
            Connection Information (c): IN IP4 192.168.2.30
            Time Description, active time (t): 0 0
            Session Attribute (a): sendrecv
            Media Description, name and address (m): audio 2224 RTP/AVP 9 0 8 18 127
            Media Attribute (a): rtpmap:9 G722/8000
            Media Attribute (a): rtpmap:0 PCMU/8000
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): rtpmap:18 G729/8000
            Media Attribute (a): fmtp:18 annexb=no
            Media Attribute (a): rtpmap:127 telephone-event/8000
Frame 7: 534 bytes on wire (4272 bits), 534 bytes captured (4272 bits)
Ethernet II, Src: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3), Dst: Polycom_68:55:6d (00:04:f2:68:55:6d)
Internet Protocol Version 4, Src: 192.168.2.1, Dst: 192.168.2.30
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol (100)
    Status-Line: SIP/2.0 100 Trying
    Message Header
        Via: SIP/2.0/UDP 192.168.2.30;branch=z9hG4bK6986a4b12DE4646;received=192.168.2.30;rport=5060
        From: "10" <sip:10@192.168.2.1>;tag=D9B2C98E-59EF4901
        To: <sip:11@192.168.2.1;user=phone>
        Call-ID: 50fbb22a-8f4bdfd-3f1393c8@192.168.2.30
        CSeq: 2 INVITE
        Server: Asterisk PBX 11.13.1~dfsg-2+deb8u2
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
        Supported: replaces, timer
        Contact: <sip:11@192.168.2.1:5060>
        Content-Length: 0
Frame 8: 42 bytes on wire (336 bits), 42 bytes captured (336 bits)
Ethernet II, Src: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3), Dst: Broadcast (ff:ff:ff:ff:ff:ff)
Address Resolution Protocol (request)
    Hardware type: Ethernet (1)
    Protocol type: IPv4 (0x0800)
    Hardware size: 6
    Protocol size: 4
    Opcode: request (1)
    Sender MAC address: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3)
    Sender IP address: 192.168.2.1
    Target MAC address: 00:00:00_00:00:00 (00:00:00:00:00:00)
    Target IP address: 192.168.2.31
Frame 9: 60 bytes on wire (480 bits), 60 bytes captured (480 bits)
Ethernet II, Src: Polycom_68:55:6e (00:04:f2:68:55:6e), Dst: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3)
Address Resolution Protocol (reply)
    Hardware type: Ethernet (1)
    Protocol type: IPv4 (0x0800)
    Hardware size: 6
    Protocol size: 4
    Opcode: reply (2)
    Sender MAC address: Polycom_68:55:6e (00:04:f2:68:55:6e)
    Sender IP address: 192.168.2.31
    Target MAC address: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3)
    Target IP address: 192.168.2.1

 

followed by invite from asterisk to the phone.

Frame 10: 1980 bytes on wire (15840 bits), 1980 bytes captured (15840 bits)
Ethernet II, Src: Giga-Byt_89:6d:b3 (74:d4:35:89:6d:b3), Dst: Polycom_68:55:6e (00:04:f2:68:55:6e)
Internet Protocol Version 4, Src: 192.168.2.1, Dst: 192.168.2.31
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:11@192.168.2.31 SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK07a6a565;rport
        Max-Forwards: 70
        From: "10" <sip:10@192.168.2.1>;tag=as4bfe3d78
        To: <sip:11@192.168.2.31>
        Contact: <sip:10@192.168.2.1:5060>
        Call-ID: 08d7a6414d054d976453b44d1eabd959@192.168.2.1:5060
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u2
        Date: Sat, 21 Jan 2017 15:54:33 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
        Supported: replaces, timer
        Content-Type: application/sdp
        Content-Length: 1371
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): root 1483002026 1483002026 IN IP4 192.168.2.1
            Session Name (s): Asterisk PBX 11.13.1~dfsg-2+deb8u2
            Connection Information (c): IN IP4 192.168.2.1
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 12734 RTP/AVP 0 4 3 3 8 8 112 5 7 18 110 97 111 9 102 115 116 117 96 100 107 108 10 118 119 127
            Media Attribute (a): rtpmap:0 PCMU/8000
            Media Attribute (a): rtpmap:4 G723/8000
            Media Attribute (a): fmtp:4 annexa=no
            Media Attribute (a): rtpmap:3 GSM/8000
            Media Attribute (a): rtpmap:3 GSM/8000
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): rtpmap:112 AAL2-G726-32/8000
            Media Attribute (a): rtpmap:5 DVI4/8000
            Media Attribute (a): rtpmap:7 LPC/8000
            Media Attribute (a): rtpmap:18 G729/8000
            Media Attribute (a): fmtp:18 annexb=no
            Media Attribute (a): rtpmap:110 speex/8000
            Media Attribute (a): rtpmap:97 iLBC/8000
            Media Attribute (a): fmtp:97 mode=30
            Media Attribute (a): rtpmap:111 G726-32/8000
            Media Attribute (a): rtpmap:9 G722/8000
            Media Attribute (a): rtpmap:102 G7221/16000
            Media Attribute (a): fmtp:102 bitrate=32000
            Media Attribute (a): rtpmap:115 G7221/32000
            Media Attribute (a): fmtp:115 bitrate=48000
            Media Attribute (a): rtpmap:116 G719/48000
            Media Attribute (a): fmtp:116 bitrate=64000
            Media Attribute (a): rtpmap:117 speex/16000
            Media Attribute (a): rtpmap:96 SILK/8000
            Media Attribute (a): fmtp:96 maxaveragebitrate=10000
            Media Attribute (a): fmtp:96 usedtx=0
            Media Attribute (a): fmtp:96 useinbandfec=1
            Media Attribute (a): rtpmap:100 SILK/12000
            Media Attribute (a): fmtp:100 maxaveragebitrate=12000
            Media Attribute (a): fmtp:100 usedtx=0
            Media Attribute (a): fmtp:100 useinbandfec=1
            Media Attribute (a): rtpmap:107 SILK/16000
            Media Attribute (a): fmtp:107 maxaveragebitrate=20000
            Media Attribute (a): fmtp:107 usedtx=0
            Media Attribute (a): fmtp:107 useinbandfec=1
            Media Attribute (a): rtpmap:108 SILK/24000
            Media Attribute (a): fmtp:108 maxaveragebitrate=30000
            Media Attribute (a): fmtp:108 usedtx=0
            Media Attribute (a): fmtp:108 useinbandfec=1
            Media Attribute (a): rtpmap:10 L16/8000
            Media Attribute (a): rtpmap:118 L16/16000
            Media Attribute (a): rtpmap:119 speex/32000
            Media Attribute (a): rtpmap:127 telephone-event/8000
            Media Attribute (a): fmtp:127 0-16
            Media Attribute (a): ptime:20
            Media Attribute (a): sendrecv

But the phone never answers. asterisk tires to send invite packets, phone never answers. I have no clue what's going on. Again, no NAT is involved, packets are on the wire. I have log level 0 enabled for SIP on the phone, it shows nothing. specifically no lines for any received packet. same thing happens if i call from "11" to "10". And both phones can call outside world. its just for some reason asterisk can't connect to the phone at all.

Please help - i have no idea what to look for... 😞

 

 

1 REPLY 1
HP Recommended

Hello lelik1,

welcome to the Polycom Community.

 

You should start using the correct and supported software for your phones and server type.


The community's VoIP FAQ contains this post here:

Jun 30, 2015 Question:What is the difference using the UC Software 4.0.0 for compatible SoundPoint or SoundStation IP Phones?

Resolution: Please always check the release Notes or

Software Version  Call Server
4.0.X SIP Only
4.1.X LYNC Only

 

Please utilize UC Software 4.0.11 and not UC Software 4.1.1

 

Once you have done this and you still have issues try here:

 

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

Resolution: Please check => here <= 

 

We would need logs from the phone end.


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.
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For HP products please check HP Support.

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