After updates (distro and freepbx) my UDP connected devices are unavailable

hi, i am using FreePBX 14.0.1.36 with asterisk 13.19.1 with pjsip endpoints. over the weekend, i applied all distro and freepbx module updates. after restarting the pbx, i noticed that the 3 polycom ip5000’s i have are showing unavailable in the “pjsip show contacts” list. in addition, they are occasionally showing up in the “blocked attacker” list (although not always). the 90+ grandstream desk phones I have are not behaving this way and the poly’s were working fine prior. I am looking at the “pjsip set logging on” and i see the initial register fails but a followup register seems to be ok. When I try a call, the call sets up, the other party answers but no audio either direction.

I am not sure where to go from here, can someone point me in the right direction…here are some logs

UPDATE: I just became aware that any phone that is udp (no encryption) configured is behaving the same…it seems to register but is show up as unavailable and cannot make or receive calls. these were all working before the updates…All my TLS connected phones work fine.

anyone have any idea’s

Initial register:

uepbx1*CLI>
<— Received SIP request (534 bytes) from UDP:172.30.2.50:5060 —>
REGISTER sip:172.30.2.1:5060 SIP/2.0

Via: SIP/2.0/UDP 172.30.2.50;branch=z9hG4bK229b41cf2F2540

From: “Phoenix Conference Room” sip:[email protected];tag=5F2FA309-6B3DB3C2

To: sip:[email protected]

CSeq: 1 REGISTER

Call-ID: [email protected]

Contact: sip:[email protected];methods=“INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER”

User-Agent: PolycomSoundStationIP-SSIP_5000-UA/4.1.1.0731

Accept-Language: en

Max-Forwards: 70

Expires: 120

Content-Length: 0

uepbx1*CLI>
<— Transmitting SIP response (516 bytes) to UDP:172.30.2.50:5060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 172.30.2.50;rport=5060;received=172.30.2.50;branch=z9hG4bK229b41cf2F2540

Call-ID: [email protected]

From: “Phoenix Conference Room” sip:[email protected];tag=5F2FA309-6B3DB3C2

To: sip:[email protected];tag=z9hG4bK229b41cf2F2540

CSeq: 1 REGISTER

WWW-Authenticate: Digest realm=“asterisk”,nonce=“1521555366/f3db9f9aedd38b761bd2e53663ab6171”,opaque=“0232a6743103aea9”,algorithm=md5,qop=“auth”

Server: FPBX-14.0.1.36(13.19.1)

Content-Length: 0

uepbx1*CLI>
<— Received SIP request (808 bytes) from UDP:172.30.2.50:5060 —>
REGISTER sip:172.30.2.1:5060 SIP/2.0

Via: SIP/2.0/UDP 172.30.2.50;branch=z9hG4bK63be8ec751E5501

From: “Phoenix Conference Room” sip:[email protected];tag=5F2FA309-6B3DB3C2

To: sip:[email protected]

CSeq: 2 REGISTER

Call-ID: [email protected]

Contact: sip:[email protected];methods=“INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER”

User-Agent: PolycomSoundStationIP-SSIP_5000-UA/4.1.1.0731

Accept-Language: en

Authorization: Digest username=“5353”, realm=“asterisk”, nonce=“1521555366/f3db9f9aedd38b761bd2e53663ab6171”, qop=auth, cnonce=“zjueJVIz3mYQTA7”, nc=00000001, opaque=“0232a6743103aea9”, uri=“sip:172.30.2.1:5060”, response=“23b42f427c2954061632412cd23bdf4e”, algorithm=MD5

Max-Forwards: 70

Expires: 120

Content-Length: 0

uepbx1*CLI>
<— Transmitting SIP response (462 bytes) to UDP:172.30.2.50:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.30.2.50;rport=5060;received=172.30.2.50;branch=z9hG4bK63be8ec751E5501

Call-ID: [email protected]

From: “Phoenix Conference Room” sip:[email protected];tag=5F2FA309-6B3DB3C2

To: sip:[email protected];tag=z9hG4bK63be8ec751E5501

CSeq: 2 REGISTER

Date: Tue, 20 Mar 2018 14:16:06 GMT

Contact: sip:[email protected]:5060;expires=119

Expires: 120

Server: FPBX-14.0.1.36(13.19.1)

Content-Length: 0

uepbx1*CLI>
<— Transmitting SIP request (420 bytes) to UDP:172.30.2.50:5060 —>
OPTIONS sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 172.30.200.1:5060;rport;branch=z9hG4bKPj9b99e675-5889-4ac9-b537-0c8712a86138

From: sip:[email protected];tag=60a1ff0f-3548-4fdc-9e0d-5cc7b3807706

To: sip:[email protected]

Contact: sip:[email protected]:5060

Call-ID: 056dd420-0497-40a2-bf14-d882a61ae6ac

CSeq: 24538 OPTIONS

Max-Forwards: 70

User-Agent: FPBX-14.0.1.36(13.19.1)

Content-Length: 0

<— Transmitting SIP request (630 bytes) to UDP:172.30.2.50:5060 —>
NOTIFY sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 172.30.200.1:5060;rport;branch=z9hG4bKPje47c0a10-b8f9-4ea2-ad6a-f34f9aba58df

From: sip:[email protected];tag=2995f3df-9e99-4e02-8277-5ec46a0d7903

To: sip:[email protected]

Contact: sip:[email protected]:5060

Call-ID: 650d92e5-1d92-46ce-ba16-05ae2a1a2e62

CSeq: 3416 NOTIFY

Subscription-State: terminated

Event: message-summary

Allow-Events: presence, dialog, message-summary, refer

Max-Forwards: 70

User-Agent: FPBX-14.0.1.36(13.19.1)

Content-Type: application/simple-message-summary

Content-Length: 48

Messages-Waiting: no

Voice-Message: 0/0 (0/0)

call attempt:

uepbx1*CLI>
<— Received SIP request (932 bytes) from UDP:172.30.2.50:5060 —>
INVITE sip:[email protected]:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 172.30.2.50;branch=z9hG4bK38dbdea3A50E2024

From: “Phoenix Conference Room” sip:[email protected];tag=B1AC897D-7F5544C6

To: sip:[email protected];user=phone

CSeq: 1 INVITE

Call-ID: [email protected]

Contact: sip:[email protected]

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER

User-Agent: PolycomSoundStationIP-SSIP_5000-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: 292

v=0

o=- 1521555777 1521555777 IN IP4 172.30.2.50

s=Polycom IP Phone

c=IN IP4 172.30.2.50

t=0 0

a=sendrecv

m=audio 2226 RTP/AVP 0 8 18 9 127

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:9 G722/8000

a=rtpmap:127 telephone-event/8000

uepbx1*CLI>
<— Transmitting SIP response (529 bytes) to UDP:172.30.2.50:5060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 172.30.2.50;rport=5060;received=172.30.2.50;branch=z9hG4bK38dbdea3A50E2024

Call-ID: [email protected]

From: “Phoenix Conference Room” sip:[email protected];tag=B1AC897D-7F5544C6

To: sip:[email protected];user=phone;tag=z9hG4bK38dbdea3A50E2024

CSeq: 1 INVITE

WWW-Authenticate: Digest realm=“asterisk”,nonce=“1521555778/1ce06a0e0a2c9043755bbc61ae1055ae”,opaque=“3b34662d7d8eeab1”,algorithm=md5,qop=“auth”

Server: FPBX-14.0.1.36(13.19.1)

Content-Length: 0

uepbx1*CLI>
<— Received SIP request (565 bytes) from UDP:172.30.2.50:5060 —>
ACK sip:[email protected]:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 172.30.2.50;branch=z9hG4bK38dbdea3A50E2024

From: “Phoenix Conference Room” sip:[email protected];tag=B1AC897D-7F5544C6

To: sip:[email protected];user=phone;tag=z9hG4bK38dbdea3A50E2024

CSeq: 1 ACK

Call-ID: [email protected]

Contact: sip:[email protected]

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER

User-Agent: PolycomSoundStationIP-SSIP_5000-UA/4.1.1.0731

Accept-Language: en

Max-Forwards: 70

Content-Length: 0

uepbx1*CLI>
<— Received SIP request (1221 bytes) from UDP:172.30.2.50:5060 —>
INVITE sip:[email protected]:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 172.30.2.50;branch=z9hG4bK8b279290C67BA2F5

From: “Phoenix Conference Room” sip:[email protected];tag=B1AC897D-7F5544C6

To: sip:[email protected];user=phone

CSeq: 2 INVITE

Call-ID: [email protected]

Contact: sip:[email protected]

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER

User-Agent: PolycomSoundStationIP-SSIP_5000-UA/4.1.1.0731

Accept-Language: en

Supported: 100rel,replaces

Allow-Events: conference,talk,hold

Authorization: Digest username=“5353”, realm=“asterisk”, nonce=“1521555778/1ce06a0e0a2c9043755bbc61ae1055ae”, qop=auth, cnonce=“fSD1YRHONCYBdFo”, nc=00000001, opaque=“3b34662d7d8eeab1”, uri=“sip:[email protected]:5060;user=phone”, response=“30c996998c2ffe50709ef86e8c6e44fd”, algorithm=MD5

Max-Forwards: 70

Content-Type: application/sdp

Content-Length: 292

v=0

o=- 1521555777 1521555777 IN IP4 172.30.2.50

s=Polycom IP Phone

c=IN IP4 172.30.2.50

t=0 0

a=sendrecv

m=audio 2226 RTP/AVP 0 8 18 9 127

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:9 G722/8000

a=rtpmap:127 telephone-event/8000

uepbx1*CLI>
<— Transmitting SIP response (348 bytes) to UDP:172.30.2.50:5060 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.30.2.50;rport=5060;received=172.30.2.50;branch=z9hG4bK8b279290C67BA2F5

Call-ID: [email protected]

From: “Phoenix Conference Room” sip:[email protected];tag=B1AC897D-7F5544C6

To: sip:[email protected];user=phone

CSeq: 2 INVITE

Server: FPBX-14.0.1.36(13.19.1)

Content-Length: 0

You haven’t got your RFC1918 address space in the trusted, or internal zone.

xrobau, currently we have the firewall disabled for testing and the problem persists…

Then it wouldn’t be showing up in the firewall blocked list. Anyway, the trace above doesn’t show any problems. One of the common issues with some phones (specifically Polycom) is that they don’t agree on Codec choices the way they should, leading to no audio. This is a common issue with PJSIP, which scrupulously adheres to the SIP RFCs.

I’d suggest you limit both the phone and the extension to only one codec (G722 only, for example), and see if that fixes your problem.

xrobau, I know you are taking a look at this with Gerardo and I wanted to make sure you had all the info. When I originally made this post, I did not know all the facts, so here is what happened.

On a sunday, i applied all system and module updates. because my phones seemed ok, i thought i was good. but i did not test well enough. what i realized the next day was that devices connected via tcp/tls (sip-tls and srtp) seemed fine (about 85 grandstream phones). I have a handful of UDP devices and these seemed not to be working. Specifically 3 polycom conference phones, a vega 50 for faxes and a valcom paging gateway.
As a troubleshooting step, i converted the polycom’s to tcp/tls and with that change, they started working fine. while the vega 50 does not support tls, it does support tcp and after changing it to tcp, it is also now working as well. both are now working with not other changes than transport.
My valcom is the real problem right now, it does not support tcp at all, so i have no choice with it and it is still not working. i have attached packet captures to the ticket but it seems to register fine, however, pjsip show contacts lists the extensions as unavailable and calling the extensions just gets you a hangup.
we currently have fail2ban and firewall disabled.

i appreciate your help

Hi tony

Rob is not currently (nor has he been) working on this issue in the internal tracker

Can I ask a favor of you. Please keep all of this dialogue in the support ticket you have opened. Putting some information here and some on the support ticket is not only confusing but slows down the process.

Thanks for your understanding.

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