Freepbx -vs- trixbox - my phones work with trixbox... not with freepbx

I’m trying to evaluate freepbx -vs- trixbox.

So far… I have installed trixbox and freepbx on a linux KVM
virtual image. I have both system installed and I can make
a SIP Phone registration using my SNOM 320 phones.

Without setting up any trunks or outbound routes, I can pick up and get a dial tone from either the freepbx connection or the trixbox connection. If I dial any local extension… that’s where things go bad for the freepbx installation.

on the freepbx installation, when I dial a local extension, I just get a fast busy signal and the phone displays: “Not acceptable here: 211” and I get a fast busy signal.

on the trixbox installation, when I dial a local extension, I
get the voice mail or number not in service message, etc.

My snom 320 log shows:
[5] 2/7/2011 09:30:07: set_audible: RP27 changed=1, update_req=0, adr=
[6] 2/7/2011 09:30:07: set_additional_addresses
[8] 2/7/2011 09:30:07: SIP: No special routing, routing to sip:[email protected];user=phone
[8] 2/7/2011 09:30:07: SIP: route pending packet 1000373/6: url ? sip:[email protected];user=phone
[8] 2/7/2011 09:30:07: SIP: route pending packet 1000373/6: udp 192.168.1.130 5060
[8] 2/7/2011 09:30:07: SIP: send Request INVITE (3c269435f3b4-jgx0qmyyqrb4/hfdt6a62wm/)
[9] 2/7/2011 09:30:07: result of get_ip_adr:192.168.1.130 192.168.1.173
[9] 2/7/2011 09:30:07: result of get_ip_adr:192.168.1.130 192.168.1.173
[5] 2/7/2011 09:30:07: Dialog 27/45 going to trying
[7] 2/7/2011 09:30:07: settings::apply_value: callrecord_dialed_time = ’ 9:30AM’, set.need_apply: 0, finished: 1, need reboot to apply: 0
[8] 2/7/2011 09:30:07: Goto State ‘Calling’ (8) from’ Edit_number’ (7), MB 0
[8] 2/7/2011 09:30:07: SIP: recvd Response 401 270771440 INVITE (3c269435f3b4-jgx0qmyyqrb4/hfdt6a62wm/as5196062a)
[9] 2/7/2011 09:30:07: SIP: clear message_repetition 1000373/3c269435f3b4-jgx0qmyyqrb4
[8] 2/7/2011 09:30:07: Routing to explicit plan udp 192.168.1.130 5060
[8] 2/7/2011 09:30:07: SIP: route pending packet 1000374/7: udp 192.168.1.130 5060
[8] 2/7/2011 09:30:07: SIP: send Request ACK (3c269435f3b4-jgx0qmyyqrb4/hfdt6a62wm/as5196062a)
[9] 2/7/2011 09:30:07: result of get_ip_adr:192.168.1.130 192.168.1.173
[9] 2/7/2011 09:30:07: result of get_ip_adr:192.168.1.130 192.168.1.173
[5] 2/7/2011 09:30:07: sip::process_auth:Match challenge for user=211, realm=asterisk
[8] 2/7/2011 09:30:07: Routing to explicit plan udp 192.168.1.130 5060
[8] 2/7/2011 09:30:07: SIP: route pending packet 1000375/8: udp 192.168.1.130 5060
[8] 2/7/2011 09:30:07: SIP: send Request INVITE (3c269435f3b4-jgx0qmyyqrb4/hfdt6a62wm/)
[9] 2/7/2011 09:30:07: result of get_ip_adr:192.168.1.130 192.168.1.173
[9] 2/7/2011 09:30:07: result of get_ip_adr:192.168.1.130 192.168.1.173
[8] 2/7/2011 09:30:08: SIP: recvd Response 401 270793616 INVITE (3c269435f3b4-jgx0qmyyqrb4/hfdt6a62wm/as5196062a)
[8] 2/7/2011 09:30:08: SIP: recvd Response 488 270793616 INVITE (3c269435f3b4-jgx0qmyyqrb4/hfdt6a62wm/as5196062a)
[9] 2/7/2011 09:30:08: SIP: clear message_repetition 1000375/3c269435f3b4-jgx0qmyyqrb4
[8] 2/7/2011 09:30:08: Routing to explicit plan udp 192.168.1.130 5060
[8] 2/7/2011 09:30:08: SIP: route pending packet 1000376/9: udp 192.168.1.130 5060
[8] 2/7/2011 09:30:08: SIP: send Request ACK (3c269435f3b4-jgx0qmyyqrb4/hfdt6a62wm/as5196062a)
[9] 2/7/2011 09:30:08: result of get_ip_adr:192.168.1.130 192.168.1.173
[9] 2/7/2011 09:30:08: result of get_ip_adr:192.168.1.130 192.168.1.173
[5] 2/7/2011 09:30:08: Dialog 27/45 going to terminated
[9] 2/7/2011 09:30:08: SIP: delete connection 27 in 120 secs
[9] 2/7/2011 09:30:08: gui::if_state(27, SIP/2.0 488 Not acceptable here)
[8] 2/7/2011 09:30:08: Goto Best State from ‘Calling’ (8), force 0
[8] 2/7/2011 09:30:08: Goto State ‘Calling’ (8) from’ Calling’ (8), MB 0
[8] 2/7/2011 09:30:08: Goto Best State from ‘Calling’ (8), force 0
[8] 2/7/2011 09:30:08: Goto State ‘Terminated’ (14) from’ Calling’ (8), MB 0
[7] 2/7/2011 09:30:08: GUI: set_led: nr 1 id 1 state 0
[9] 2/7/2011 09:30:08: SIP: delete connection 27 in 0 secs
[9] 2/7/2011 09:30:08: Timer: Registering with timeout of 0 ms
[9] 2/7/2011 09:30:08: SIP: delete connection 27 in 0 secs

… but I don’t know what much of any of that means…

I’d really like to continue to evaluate freepbx… it seems to be updated more often than trixbox and tribox hasn’t been updated in over a year… but I can actually make calls with trixbox and our snom voip phones, it… puts my freepbx eval as a huge disadvantage…

I can use a soft phone ( twinkle on linux ) and make a call using freepbx… it’s the real, voip, hardware phone that I am having issues with… on freepbx… not trixbox…

Would appreciate any help I can get. Thanks - jack

The SNOM’s set Secure RTP by default. Disable secure RTP on the phone and you will be good to go.

that did the trick. Thanks. I set “RTP Encryption:” on the snom phone to off ( it was on ) and now the phone talks to freepbx just fine. Thanks for the quick response.

jack

Yes, the older version of Asterisk did not support SRTP, Asterisk 1.8 does and that was the source of your problem.

I see several areas where this is referenced
Snom 300: Symmetrical RTP on/off in Each Identity.
Under Advanced: RTCP Support: off and RTP Keepalive:
I have tried shutting these off - I’m still losing registration.