service provider is not down and the firewall is not an issue. I completely disabled the firewall of the router and still didn't help. I tried to connect to sip provider via soft phone it is connecting and I am making phone calls from the some network environment. I also tried to register another sip account from a different provider. It is not also connecting.
In mean time I setup a snomone to mac book pro with lion os but not server. I have just introduced the sip registrations again got 408 but re started the snomone it is 200ok for two sip registration. Again I set up simple dial plan 200 *. Still I cannot make any call. Below is the log file of this call.
[8] 2012/04/03 01:24:58: Allocating call port 8, SIP call id c5705d42850d8995
[1] 2012/04/03 01:24:58: UDP: TOS could not be set
[1] 2012/04/03 01:24:58: Last message repeated 2 times
[8] 2012/04/03 01:24:58: Could not find a trunk (3 trunks)
[8] 2012/04/03 01:24:58: Tagging request with existing tag
[6] 2012/04/03 01:24:58: Sending RTP for c5705d42850d8995 to 192.168.1.2:10000, codec not set yet
[8] 2012/04/03 01:24:58: Incoming call: Request URI sip:+40213674050@192.168.1.5, To is <sip:+40213674050@192.168.1.5>
[8] 2012/04/03 01:24:58: Call from an user 40
[8] 2012/04/03 01:24:58: To is <sip:+40213674050@192.168.1.5>, user 0, domain 1
[8] 2012/04/03 01:24:58: From user 40
[8] 2012/04/03 01:24:58: Set the To domain based on From user 40@localhost
[8] 2012/04/03 01:24:58: Call state for call object 5: idle
[7] 2012/04/03 01:24:58: set_codecs: for c5705d42850d8995 codecs "", codec_preference count 6
[5] 2012/04/03 01:24:58: Dialplan "Standard Dialplan": Match +40213674050@192.168.1.5 to <sip:+40213674050@sip.didlogic.com;user=phone> on trunk didlogic
[8] 2012/04/03 01:24:58: Play audio_moh/noise.wav
[8] 2012/04/03 01:24:58: Allocating call port 9, SIP call id e89574bd@pbx
[1] 2012/04/03 01:24:58: UDP: TOS could not be set
[1] 2012/04/03 01:24:58: Last message repeated 2 times
[7] 2012/04/03 01:24:58: set_codecs: for e89574bd@pbx codecs "", codec_preference count 6
[8] 2012/04/03 01:24:58: call port 9: state code from 0 to 100
[8] 2012/04/03 01:24:58: DNS: Request sip.didlogic.com from server 192.168.1.1
[8] 2012/04/03 01:24:58: call port 8: state code from 0 to 183
[6] 2012/04/03 01:24:58: Codec pcmu/8000 is chosen for call id c5705d42850d8995
[8] 2012/04/03 01:24:59: DNS: SRV _sip._tcp.sip.didlogic.com expired
[8] 2012/04/03 01:25:08: DNS: Add NAPTR sip.didlogic.com (ttl=60)
[8] 2012/04/03 01:25:08: DNS: Request _sips._tcp.sip.didlogic.com from server 192.168.1.1
[8] 2012/04/03 01:25:08: DNS: SRV _sip._udp.sip.didlogic.com expired
[8] 2012/04/03 01:25:13: call port 9: state code from 100 to 486
[8] 2012/04/03 01:25:13: Remove leg 9: call port 8, SIP call id c5705d42850d8995
[8] 2012/04/03 01:25:13: DNS: we have already a request pending for this address SRV _sips._tcp.sip.didlogic.com
[8] 2012/04/03 01:25:13: Clearing call port 8, SIP call id c5705d42850d8995
[8] 2012/04/03 01:25:13: Hangup: Call 8 not found
Again to test if the sip registration is working on mac book pro. I called the sip aacount and snom one managed to get the call and divert it to the assigned account. Below the log of this incoming call.
[8] 2012/04/03 01:38:50: Incoming call: Request URI sip:38353@192.168.1.5:5060;transport=udp;line=eccbc87e, To is <sip:38353@192.168.1.5:5060;transport=udp;line=eccbc87e>
[8] 2012/04/03 01:38:50: Call from a trunk 3
[8] 2012/04/03 01:38:50: Trunk didlogic@localhost has country code not set, area code not set
[8] 2012/04/03 01:38:50: To is <sip:38353@192.168.1.5:5060;transport=udp;line=eccbc87e;user=phone>, user 0, domain 1
[8] 2012/04/03 01:38:50: Send call to extension ERE returned 40
[5] 2012/04/03 01:38:50: Domain trunk didlogic@localhost sends call to 40 in domain localhost
[8] 2012/04/03 01:38:50: Set the To domain based on To user 40@localhost
[8] 2012/04/03 01:38:50: Call state for call object 6: idle
[7] 2012/04/03 01:38:50: set_codecs: for 389573436816637c70d25809302cfbae@178.63.100.24 codecs "", codec_preference count 6
[8] 2012/04/03 01:38:50: Call state for call object 6: alerting
[8] 2012/04/03 01:38:50: Play audio_moh/noise.wav
[8] 2012/04/03 01:38:50: Allocating call port 11, SIP call id 97e6f51e@pbx
[1] 2012/04/03 01:38:50: UDP: TOS could not be set
[1] 2012/04/03 01:38:50: Last message repeated 2 times
[7] 2012/04/03 01:38:50: set_codecs: for 97e6f51e@pbx codecs "", codec_preference count 6
[8] 2012/04/03 01:38:50: call port 11: state code from 0 to 100
[8] 2012/04/03 01:38:50: call port 10: state code from 0 to 100
[7] 2012/04/03 01:38:50: Set packet length to 20
[8] 2012/04/03 01:38:50: Play audio_en/ringback.wav
[8] 2012/04/03 01:38:50: call port 10: state code from 100 to 183
[6] 2012/04/03 01:38:50: Codec pcmu/8000 is chosen for call id 389573436816637c70d25809302cfbae@178.63.100.24
[8] 2012/04/03 01:38:59: DNS: SRV _sip._tcp.sip.didlogic.com expired
[7] 2012/04/03 01:39:05: Call 97e6f51e@pbx: Clear last INVITE
[6] 2012/04/03 01:39:05: Codec pcmu/8000 is chosen for call id 97e6f51e@pbx
[6] 2012/04/03 01:39:05: Sending RTP for 97e6f51e@pbx to 192.168.1.2:10000, codec pcmu/8000
[7] 2012/04/03 01:39:05: Determine pass-through mode after receiving response
[8] 2012/04/03 01:39:05: Call state for call object 6: connected
[8] 2012/04/03 01:39:05: call port 11: state code from 100 to 200
[8] 2012/04/03 01:39:05: call port 10: state code from 183 to 200
[7] 2012/04/03 01:39:05: 389573436816637c70d25809302cfbae@178.63.100.24: RTP pass-through mode
[7] 2012/04/03 01:39:05: 97e6f51e@pbx: RTP pass-through mode
[5] 2012/04/03 01:39:06: Tuning to new SSRC
[8] 2012/04/03 01:39:09: DNS: SRV _sip._udp.sip.didlogic.com expired
[7] 2012/04/03 01:39:12: 389573436816637c70d25809302cfbae@178.63.100.24: Media-aware pass-through mode
[8] 2012/04/03 01:39:12: Clearing call port 11, SIP call id 97e6f51e@pbx
[8] 2012/04/03 01:39:12: call port 10: state code from 200 to 486
[8] 2012/04/03 01:39:12: Remove leg 12: call port 11, SIP call id 97e6f51e@pbx
[8] 2012/04/03 01:39:12: Hangup: Call 11 not found
[8] 2012/04/03 01:39:12: Last message repeated 2 times
[7] 2012/04/03 01:39:12: Call 389573436816637c70d25809302cfbae@178.63.100.24: Clear last request
[5] 2012/04/03 01:39:12: BYE Response: Terminate 389573436816637c70d25809302cfbae@178.63.100.24
[8] 2012/04/03 01:39:12: Clearing call port 10, SIP call id 389573436816637c70d25809302cfbae@178.63.100.24
[8] 2012/04/03 01:39:12: Remove leg 11: call port 10, SIP call id 389573436816637c70d25809302cfbae@178.63.100.24
to make sure that same applies to second sip registration as well I set up the same simple dial plan and result was the same I couldn't dial out. Again the log is below.
8] 2012/04/03 01:44:00: DNS: Add SRV _sips._tcp.sip.mydivert.com (ttl=60)
[8] 2012/04/03 01:44:00: DNS: Request _sip._tcp.sip.mydivert.com from server 192.168.1.1
[8] 2012/04/03 01:44:09: Allocating call port 12, SIP call id 8ea1c994b396a16e
[1] 2012/04/03 01:44:09: UDP: TOS could not be set
[1] 2012/04/03 01:44:09: Last message repeated 2 times
[8] 2012/04/03 01:44:09: Could not find a trunk (3 trunks)
[8] 2012/04/03 01:44:09: Tagging request with existing tag
[6] 2012/04/03 01:44:09: Sending RTP for 8ea1c994b396a16e to 192.168.1.2:10000, codec not set yet
[8] 2012/04/03 01:44:09: Incoming call: Request URI sip:+40213674050@192.168.1.5, To is <sip:+40213674050@192.168.1.5>
[8] 2012/04/03 01:44:09: Call from an user 40
[8] 2012/04/03 01:44:09: To is <sip:+40213674050@192.168.1.5>, user 0, domain 1
[8] 2012/04/03 01:44:09: From user 40
[8] 2012/04/03 01:44:09: Set the To domain based on From user 40@localhost
[8] 2012/04/03 01:44:09: Call state for call object 7: idle
[7] 2012/04/03 01:44:09: set_codecs: for 8ea1c994b396a16e codecs "", codec_preference count 6
[5] 2012/04/03 01:44:09: Dialplan "Standard Dialplan": Match +40213674050@192.168.1.5 to <sip:+40213674050@sip.mydivert.com;user=phone> on trunk mydivert
[8] 2012/04/03 01:44:09: Play audio_moh/noise.wav
[8] 2012/04/03 01:44:09: Allocating call port 13, SIP call id cbba4240@pbx
[1] 2012/04/03 01:44:09: UDP: TOS could not be set
[1] 2012/04/03 01:44:09: Last message repeated 2 times
[7] 2012/04/03 01:44:09: set_codecs: for cbba4240@pbx codecs "", codec_preference count 6
[8] 2012/04/03 01:44:09: call port 13: state code from 0 to 100
[8] 2012/04/03 01:44:09: DNS: we have already a request pending for this address SRV _sip._tcp.sip.mydivert.com
[8] 2012/04/03 01:44:09: call port 12: state code from 0 to 183
[6] 2012/04/03 01:44:09: Codec pcmu/8000 is chosen for call id 8ea1c994b396a16e
[8] 2012/04/03 01:44:10: DNS: Add SRV _sip._tcp.sip.mydivert.com (ttl=60)
[8] 2012/04/03 01:44:10: DNS: Request _sip._udp.sip.mydivert.com from server 192.168.1.1
[8] 2012/04/03 01:44:10: DNS: we have already a request pending for this address SRV _sip._udp.sip.mydivert.com
[8] 2012/04/03 01:44:20: DNS: Add SRV _sip._udp.sip.mydivert.com (ttl=60)
[4] 2012/04/03 01:44:20: Could not find packet with number 167
[8] 2012/04/03 01:44:20: DNS: Request sip.mydivert.com from server 192.168.1.1
[8] 2012/04/03 01:44:30: DNS: Add AAAA sip.mydivert.com (ttl=60)
[8] 2012/04/03 01:44:30: DNS: Request sip.mydivert.com from server 192.168.1.1
[8] 2012/04/03 01:44:30: DNS: Add A sip.mydivert.com 78.46.43.9 (ttl=900)
[7] 2012/04/03 01:44:30: Call cbba4240@pbx: Clear last INVITE
[5] 2012/04/03 01:44:30: INVITE Response 603 Declined: Terminate cbba4240@pbx
[8] 2012/04/03 01:44:30: Clearing call port 13, SIP call id cbba4240@pbx
[8] 2012/04/03 01:44:30: call port 12: state code from 183 to 603
[8] 2012/04/03 01:44:30: Remove leg 14: call port 13, SIP call id cbba4240@pbx
[8] 2012/04/03 01:44:30: Hangup: Call 13 not found
[8] 2012/04/03 01:44:30: Clearing call port 12, SIP call id 8ea1c994b396a16e
[8] 2012/04/03 01:44:30: Remove leg 13: call port 12, SIP call id 8ea1c994b396a16e
[8] 2012/04/03 01:44:50: DNS: NAPTR sip.mydivert.com expired
do you have a recommendation for the issues. with the above work I think it is very clear that sip providers are not the issue. There is something wrong in snom one side for sip registration to get 200 ok and use dial plan and to make outbound calls