Endpoint over vlan call drops after 30s, no audio

I’ve recently changed a networking change on my PBXACT UC60, bringing up a second network interface specifically for wan and making it the default gateway, adding a static route for the vlan’d segment as shown below.
After this change, endpoints (Fanvil door phones connected via pjsip) on a vlan no longer work.
All endpoints on vpn (internal openvpn), and on the local network (pjsip sangoma phones) work fine.
I’ve checked the sip alg is disabled on the routing hardware.

When making a call to or from a set of endpoints on the vlan, no audio is heard, call is dropped after ~30s.

Traffic seems to flow unimpeded between pbxact and the endpoints in question, and they worked fine when the interface they communicate over was also the default gateway. Ping works, signaling from phone->pbxact works (call is built and endpoint rung). I don’t have audio from vlan’d phone -> pbxact, It’s hard to test in the other direction as I’m 1400 miles from the door phone.

[[email protected] ~]# ip route
default via <WAN IP REMOVED> dev eth1
10.2.0.0/24 dev eth0 proto kernel scope link src 10.2.0.20
10.2.71.0/24 via 10.2.0.1 dev eth0
<additional routes removed for clarity>

Like I said, ping works so I suspect the routing is correct.
Attached is a PJSIP log for a call (vlan endpoint -> working endpoint)

PJSIP Logging Enabled for host: 10.2.71.101
<--- Received SIP request (1138 bytes) from UDP:10.2.71.101:5525 --->
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.2.71.101:5525;branch=z9hG4bK1285014997580121095;rport
From: 2301 <sip:[email protected]:5060>;tag=70510154
To: "1020" <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5525>
Max-Forwards: 70
Supported: replaces, join, path, 100rel
User-Agent: Fanvil i31S 2.6.1.6726 0c383e20cecd
P-Early-Media: supported
Allow-Events: talk,hold
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 514

v=0
o=sdp_admin 250703451 99731344 IN IP4 10.2.71.101
s=A conversation
c=IN IP4 10.2.71.101
t=0 0
m=audio 10204 RTP/AVP 9 0 8 18 102 4 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 G726-32/8000
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 10206 RTP/AVP 117
b=AS:2000
a=rtpmap:117 H264/90000
a=fmtp:117 profile-level-id=42801f; max-br=2000
a=sendonly

<--- Transmitting SIP response (517 bytes) to UDP:10.2.71.101:5525 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.2.71.101:5525;rport=5525;received=10.2.71.101;branch=z9hG4bK1285014997580121095
Call-ID: [email protected]
From: "2301" <sip:[email protected]>;tag=70510154
To: "1020" <sip:[email protected];user=phone>;tag=z9hG4bK1285014997580121095
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1612453924/4742162a099f622b1e65101485d1aab3",opaque="75b6ef091dbcfd08",algo                                                     rithm=md5,qop="auth"
Server: Asterisk PBX 15.7.4
Content-Length:  0


<--- Received SIP request (434 bytes) from UDP:10.2.71.101:5525 --->
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.2.71.101:5525;branch=z9hG4bK1285014997580121095;rport
From: "2301" <sip:[email protected]>;tag=70510154
To: "1020" <sip:[email protected];user=phone>;tag=z9hG4bK1285014997580121095
Call-ID: [email protected]
CSeq: 1 ACK
Contact: <sip:[email protected]:5525>
User-Agent: Fanvil i31S 2.6.1.6726 0c383e20cecd
Max-Forwards: 70
Content-Length: 0


<--- Received SIP request (1412 bytes) from UDP:10.2.71.101:5525 --->
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.2.71.101:5525;branch=z9hG4bK96172124996206708;rport
From: 2301 <sip:[email protected]:5060>;tag=70510154
To: "1020" <sip:[email protected];user=phone>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:5525>
Authorization: Digest username="2301", realm="asterisk", nonce="1612453924/4742162a099f622b1e65101485d1aab3", uri="sip:[email protected]                                                     10.2.0.20;user=phone", response="9025afa484349f6bd6cd96903f4919fa", algorithm=MD5, cnonce="ebcb82d2", opaque="75b6ef091dbcfd                                                     08", qop=auth, nc=00101019
Max-Forwards: 70
Supported: replaces, join, path, 100rel
User-Agent: Fanvil i31S 2.6.1.6726 0c383e20cecd
P-Early-Media: supported
Allow-Events: talk,hold
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 514

v=0
o=sdp_admin 250703451 99731344 IN IP4 10.2.71.101
s=A conversation
c=IN IP4 10.2.71.101
t=0 0
m=audio 10204 RTP/AVP 9 0 8 18 102 4 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 G726-32/8000
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 10206 RTP/AVP 117
b=AS:2000
a=rtpmap:117 H264/90000
a=fmtp:117 profile-level-id=42801f; max-br=2000
a=sendonly

<--- Transmitting SIP response (332 bytes) to UDP:10.2.71.101:5525 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.71.101:5525;rport=5525;received=10.2.71.101;branch=z9hG4bK96172124996206708
Call-ID: [email protected]
From: "2301" <sip:[email protected]>;tag=70510154
To: "1020" <sip:[email protected];user=phone>
CSeq: 2 INVITE
Server: Asterisk PBX 15.7.4
Content-Length:  0


<--- Transmitting SIP response (587 bytes) to UDP:10.2.71.101:5525 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.2.71.101:5525;rport=5525;received=10.2.71.101;branch=z9hG4bK96172124996206708
Call-ID: [email protected]
From: "2301" <sip:[email protected]>;tag=70510154
To: "1020" <sip:[email protected];user=phone>;tag=0023f880-8cf0-4c5e-978b-abff033cac2d
CSeq: 2 INVITE
Server: Asterisk PBX 15.7.4
Contact: <sip:170.249.164.98:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
P-Asserted-Identity: "[email protected]" <sip:[email protected];user=phone>
Content-Length:  0


<--- Transmitting SIP response (587 bytes) to UDP:10.2.71.101:5525 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.2.71.101:5525;rport=5525;received=10.2.71.101;branch=z9hG4bK96172124996206708
Call-ID: [email protected]
From: "2301" <sip:[email protected]>;tag=70510154
To: "1020" <sip:[email protected];user=phone>;tag=0023f880-8cf0-4c5e-978b-abff033cac2d
CSeq: 2 INVITE
Server: Asterisk PBX 15.7.4
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: <sip:170.249.164.98:5060>
P-Asserted-Identity: "[email protected]" <sip:[email protected];user=phone>
Content-Length:  0


<--- Transmitting SIP response (1004 bytes) to UDP:10.2.71.101:5525 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.71.101:5525;rport=5525;received=10.2.71.101;branch=z9hG4bK96172124996206708
Call-ID: 211771421625[email protected]
From: "2301" <sip:[email protected]>;tag=70510154
To: "1020" <sip:[email protected];user=phone>;tag=0023f880-8cf0-4c5e-978b-abff033cac2d
CSeq: 2 INVITE
Server: Asterisk PBX 15.7.4
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: <sip:170.249.164.98:5060>
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: "[email protected]" <sip:[email protected];user=phone>
Content-Type: application/sdp
Content-Length:   340

v=0
o=- 250703451 99731346 IN IP4 170.249.164.98
s=Asterisk
c=IN IP4 170.249.164.98
t=0 0
m=audio 18568 RTP/AVP 0 8 9 102 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 0 RTP/AVP 117

<--- Received SIP request (436 bytes) from UDP:10.2.71.101:5525 --->
ACK sip:170.249.164.98:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.71.101:5525;branch=z9hG4bK17595203402084824447
From: 2301 <sip:[email protected]:5060>;tag=70510154
To: "1020" <sip:[email protected];user=phone>;tag=0023f880-8cf0-4c5e-978b-abff033cac2d
Call-ID: [email protected]
CSeq: 2 ACK
Contact: <sip:[email protected]:5525>
Max-Forwards: 70
User-Agent: Fanvil i31S 2.6.1.6726 0c383e20cecd
Content-Length: 0


<--- Transmitting SIP request (419 bytes) to UDP:10.2.71.101:5525 --->
OPTIONS sip:[email protected]:5525 SIP/2.0
Via: SIP/2.0/UDP 170.249.164.98:5060;rport;branch=z9hG4bKPj85b396ac-afb6-44c8-8d72-3a52d90dc015
From: <sip:[email protected]>;tag=c1f89575-1144-4c15-8b30-f8770046f190
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 88c81630-1fd1-4c67-b236-63c37a003b41
CSeq: 32635 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.7.4
Content-Length:  0


<--- Received SIP response (543 bytes) from UDP:10.2.71.101:1287 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 170.249.164.98:5060;rport=5060;branch=z9hG4bKPj85b396ac-afb6-44c8-8d72-3a52d90dc015
From: <sip:[email protected]>;tag=c1f89575-1144-4c15-8b30-f8770046f190
To: <sip:[email protected]>;tag=4548823
Call-ID: 88c81630-1fd1-4c67-b236-63c37a003b41
CSeq: 32635 OPTIONS
Contact: <sip:[email protected]:5525>
Supported: 100rel, replaces, timer
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Accept: application/sdp, message/sipfrag, application/dtmf-relay
Content-Length: 0


[2021-02-04 09:52:38] NOTICE[9148]: res_pjsip_sdp_rtp.c:124 rtp_check_timeout: Disconnecting channel 'PJSIP/2301-00004dc8' for lack of RTP activity in 34 seconds
<--- Transmitting SIP request (436 bytes) to UDP:10.2.71.101:5525 --->
BYE sip:[email protected]:5525 SIP/2.0
Via: SIP/2.0/UDP 170.249.164.98:5060;rport;branch=z9hG4bKPj515468b8-948d-46d0-999b-bc17dc3a31c9
From: "1020" <sip:[email protected];user=phone>;tag=0023f880-8cf0-4c5e-978b-abff033cac2d
To: "2301" <sip:[email protected]>;tag=70510154
Call-ID: [email protected]
CSeq: 8683 BYE
Reason: Q.850;cause=44
Max-Forwards: 70
User-Agent: Asterisk PBX 15.7.4
Content-Length:  0


<--- Received SIP response (391 bytes) from UDP:10.2.71.101:1287 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 170.249.164.98:5060;rport=5060;branch=z9hG4bKPj515468b8-948d-46d0-999b-bc17dc3a31c9
From: "1020" <sip:[email protected];user=phone>;tag=0023f880-8cf0-4c5e-978b-abff033cac2d
To: "2301" <sip:[email protected]>;tag=70510154
Call-ID: [email protected]
CSeq: 8683 BYE
User-Agent: Fanvil i31S 2.6.1.6726 0c383e20cecd
Content-Length: 0


<--- Received SIP request (372 bytes) from UDP:10.2.71.101:5525 --->
OPTIONS sip:10.2.0.20:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.71.101:5525;branch=z9hG4bK22347441949972251;rport
From: 2301 <sip:[email protected]:5060>;tag=2974512255
To: <sip:10.2.0.20:5060>
Call-ID: [email protected]
CSeq: 1 OPTIONS
Max-Forwards: 70
User-Agent: Fanvil i31S 2.6.1.6726 0c383e20cecd
Accept: application/sdp
Content-Length: 0


<--- Transmitting SIP response (493 bytes) to UDP:10.2.71.101:5525 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.2.71.101:5525;rport=5525;received=10.2.71.101;branch=z9hG4bK22347441949972251
Call-ID: [email protected]
From: "2301" <sip:[email protected]>;tag=2974512255
To: <sip:10.2.0.20>;tag=z9hG4bK22347441949972251
CSeq: 1 OPTIONS
WWW-Authenticate: Digest realm="asterisk",nonce="1612453965/24e83b144e74b8e4c0187a22b2c5cef1",opaque="284f6b616cce759d",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.7.4
Content-Length:  0


<--- Received SIP request (636 bytes) from UDP:10.2.71.101:5525 --->
OPTIONS sip:10.2.0.20:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.71.101:5525;branch=z9hG4bK4863923670420406;rport
From: 2301 <sip:[email protected]:5060>;tag=2974512255
To: <sip:10.2.0.20:5060>
Call-ID: [email protected]
CSeq: 2 OPTIONS
Authorization: Digest username="2301", realm="asterisk", nonce="1612453965/24e83b144e74b8e4c0187a22b2c5cef1", uri="sip:10.2.0.20:5060", response="8db401daac0adc3ed1f88b95f30ab6c7", algorithm=MD5, cnonce="ebcb82d2", opaque="284f6b616cce759d", qop=auth, nc=00101020
Max-Forwards: 70
User-Agent: Fanvil i31S 2.6.1.6726 0c383e20cecd
Accept: application/sdp
Content-Length: 0


<--- Transmitting SIP response (822 bytes) to UDP:10.2.71.101:5525 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.71.101:5525;rport=5525;received=10.2.71.101;branch=z9hG4bK4863923670420406
Call-ID: [email protected]
From: "2301" <sip:[email protected]>;tag=2974512255
To: <sip:10.2.0.20>;tag=z9hG4bK4863923670420406
CSeq: 2 OPTIONS
Accept: application/sdp, application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, message/sipfrag;version=2.0
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: text/plain
Accept-Language: en
Server: Asterisk PBX 15.7.4
Content-Length:  0

In the 200 OK, Asterisk sent
Contact: <sip:170.249.xx.xx:5060>;
so it didn’t know that 10.2.71.x is “local”.

In Asterisk SIP Settings, add 10.2.71.0 / 24 to Local Networks, Submit, Apply Config and then you must restart (not just reload) Asterisk for the new settings to take effect.

If you still have trouble, post another SIP trace.

1 Like

That was it.
Added the network to the local networks in SIP settings, applied and issued core restart. :100:

Thanks for the help!

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