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All my SoundPoint IP phones stopped receiving calls

January 22, 2017, 2:43 pm
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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... :(

 

 

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