FOOD FOR THOUGHT no SIP phones can register IncrediblePBX on PI???

[bofh42]

Hi,

I'm setting up an "Incredible PBX 12.0.70" on a Raspberry PI. The Asterisk Version is
Asterisk (Ver. 13.6.0)
all updates done

Most things work as it seems:
- connection to 3 SIP trunks OK
- configuration reports no error
- but not a single phone (extension) can register

The setup uses chan_sip

all clients (phones) (Cisco ATA boxes, Zoiper softphone) are in the same LAN.
they try to register, I do see that in tshark (IP traffic dump)
377 813.266049 10.80.0.6 -> 192.168.80.12 SIP 588 Request: REGISTER sip:192.168.80.12:5060;transport=UDP (1 binding) |
378 817.266488 10.80.0.6 -> 192.168.80.12 SIP 588 Request: REGISTER sip:192.168.80.12:5060;transport=UDP (1 binding) |
379 820.840344 10.80.0.14 -> 192.168.80.12 UDP 83 Source port: 58754 Destination port: 5060
380 820.880758 10.80.0.14 -> 192.168.80.12 UDP 83 Source port: 58754 Destination port: 5060
381 820.921599 10.80.0.14 -> 192.168.80.12 UDP 83 Source port: 58754 Destination port: 5060
382 821.266095 10.80.0.6 -> 192.168.80.12 SIP 588 Request: REGISTER sip:192.168.80.12:5060;transport=UDP (1 binding) |
383 827.111068 10.80.0.14 -> 192.168.80.12 UDP 83 Source port: 58754 Destination port: 5060
384 827.111554 10.80.0.14 -> 192.168.80.12 SIP 590 Request: REGISTER sip:192.168.80.12:5060;transport=UDP (1 binding) |
385 828.012808 192.168.80.162 -> 192.168.80.12 SIP 640 Request: REGISTER sip:192.168.80.12 (1 binding) |
386 828.515996 192.168.80.162 -> 192.168.80.12 SIP 640 Request: REGISTER sip:192.168.80.12 (1 binding) |
387 829.512108 192.168.80.162 -> 192.168.80.12 SIP 640 Request: REGISTER sip:192.168.80.12 (1 binding) |
388 830.858420 10.80.0.14 -> 192.168.80.12 SIP 590 Request: REGISTER sip:192.168.80.12:5060;transport=UDP (1 binding) |
389 831.516077 192.168.80.162 -> 192.168.80.12 SIP 640 Request: REGISTER sip:192.168.80.12 (1 binding) |
390 834.319841 10.80.0.14 -> 192.168.80.12 SIP 590 Request: REGISTER sip:192.168.80.12:5060;transport=UDP (1 binding) |
391 835.512207 192.168.80.162 -> 192.168.80.12 SIP 640 Request: REGISTER sip:192.168.80.12 (1 binding) |
392 839.520399 192.168.80.162 -> 192.168.80.12 SIP 640 Request: REGISTER sip:192.168.80.12 (1 binding) |
393 841.824889 10.80.0.14 -> 192.168.80.12 UDP 83 Source port: 58754 Destination port: 5060
394 841.825396 10.80.0.14 -> 192.168.80.12 SIP 590 Request: REGISTER sip:192.168.80.12:5060;transport=UDP (1 binding) |
395 843.516291 192.168.80.162 -> 192.168.80.12 SIP 640 Request: REGISTER sip:192.168.80.12 (1 binding) |

but they always get a timeout (NOT a reject!)

iptables -L -n shows that there is nothing blocked:
ACCEPT all -- 192.168.0.0/16 0.0.0.0/0

but, even with
asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvv
and
sip set logger on

NOTHING shows up in ANY log: not in any asterisk-related log, not in freepbx log, not in /var/log/messages ....

it seems like the SIP REGISTER request just disappear

Where can I look further?

What might be the reason?


Asterisk (Ver. 13.6.0): Summary
Summary
Asterisk System uptime: 6 weeks, 3 days, 3 hours, 23 minutes, 5 seconds
Last reload: 27 minutes, 26 seconds
Active SIP Channel(s): 0 Active IAX2 Channel(s): 0
Sip Registry: 6 IAX2 Registry: 2
Sip Peers:
Online: 0
Online-Unmonitored: 5
Offline: 10
Offline-Unmonitored: 0 IAX2 Peers:
Online: 4
Offline: 0
Unmonitored: 0

Asterisk (Ver. 13.6.0): Peers
Chan_Sip Peers

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
6000 (Unspecified) D Yes Yes A 0 UNKNOWN
6001 (Unspecified) D Yes Yes A 0 UNKNOWN
6002 (Unspecified) D Yes Yes A 0 UNKNOWN
6003 (Unspecified) D Yes Yes A 0 UNKNOWN
6005 (Unspecified) D Yes Yes A 0 UNKNOWN
6006 (Unspecified) D Yes Yes A 0 UNKNOWN
6007 (Unspecified) D Yes Yes A 0 UNKNOWN
6008 (Unspecified) D Yes Yes A 0 UNKNOWN
6009 (Unspecified) D Yes Yes A 0 UNKNOWN
701 (Unspecified) D Yes Yes A 0 UNKNOWN
Berofix/berofixpi 192.168.80.160 Yes Yes 5060 Unmonitored
FreeVoipDial/bofh42 77.72.174.128 Yes Yes 5060 Unmonitored
PersonalVoiPSip1/504611 46.182.250.46 Yes Yes 5060 Unmonitored
PersonalVoipConf/504613 46.182.250.46 Yes Yes 5060 Unmonitored
PersonalVoipSIP2/504612 46.182.250.46 Yes Yes 5060 Unmonitored
15 sip peers [Monitored: 0 online, 10 offline Unmonitored: 5 online, 0 offline]

IAX2 Peers

Name/Username Host Mask Port Status Description
iax-fax0 127.0.0.1 (D) 255.255.255.255 4570 OK (1 ms)
iax-fax1 127.0.0.1 (D) 255.255.255.255 4571 OK (2 ms)
iax-fax2 127.0.0.1 (D) 255.255.255.255 4572 OK (1 ms)
iax-fax3 127.0.0.1 (D) 255.255.255.255 4573 OK (1 ms)
4 iax2 peers [4 online, 0 offline, 0 unmonitored]

 

[bofh42]

Connectivity checked - OK
routing checked - OK (it's the same subnet anyway)
iptables checked - OK (same situation with all rules flushed)
both tried can_sip and PJSIP - same result - Bummer!
SIP REGISTER request packets arrive (sngrep an tshark) - OK
asterisk ist listening on 5060 (lsof and netstat checked) - OK

nothing shows up in asterisk logs or in asterisk CLI - Bummer!

anyone out there having any idea where I can start searching (setting, debug options, tracing, ....)?

 

[bofh42]

and even a debug level of 9 in Asterisk CLI and "sip set debug on" will produce ANY output.
everything else (incl. iax2) ist very vebose, but NOTHING from chan_sip or pjsip... but they run, as sip show peers shows the trunks being connected

?????????????????

 

[wardmundy]

Neither chan_sip nor pjsip is listening on UDP 83. That's the SIP port your phone is using.

 

[bofh42]

That's not the point, sorry:

sngrep - SIP messages flow viewer
Current Mode: Online [eth0] Dialogs: 7
Match Expression: BPF Filter:
Display Filter:
^Idx Method SIP From SIP To Msgs Source Destination Call State
[ ] 1 REGISTER [email protected] [email protected] 33 192.168.80.6:46431 192.168.80.12:5060
[ ] 2 REGISTER [email protected]:5060 [email protected]:5060 23 192.168.80.144:36181 192.168.80.12:5060
[ ] 3 OPTIONS [email protected] [email protected] 2 192.168.80.12:5060 192.168.80.160:5060
[ ] 4 REGISTER [email protected]:5060 [email protected]:5060 33 192.168.80.31:48077 192.168.80.12:5060
[ ] 5 REGISTER [email protected] [email protected] 28 192.168.80.162:5061 192.168.80.12:5060
[ ] 6 OPTIONS [email protected] [email protected] 2 192.168.80.12:5060 192.168.80.160:5060
[ ] 7 OPTIONS [email protected] [email protected] 2 192.168.80.12:5060 192.168.80.160:5060


and still NOTHING shows up in any log

besides: where do you see port 83??????
it clear says: Destination Port: 5060 ?????

 

[bofh42]

Well ..

adding the SIP localnet setting for the local networks 192.168.80.0/255.255.255.0 (LAN) and 10.8.80.0/255.255.255.0 (VPN) seems to do the trick... but

the real tricky part is: SAVE/SUBMIT it BUT DO NOT "apply config" after that in FreePBX WebGUI.

instead: reboot the whole machine ...

the FreePBX GUI still has "apply config" in RED, but do not do that! that destroys the SIP settings again!

this seems very very strange!