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INVITE Response 603 declined problem

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For a whiile I am trying to setup snomone to mac os lion with server. For a long time the trunks that I have configured was not working. I found out to change the ports to different ones. I have done that so the ports are ok. But still trunks were not registering. I installed the software once again it is registering and getting 200 ok. I just created new simple dial plan which is * . when I dialed a full number wit country and area codes and two 00 at the beginning I couldn't dial out. ı put the log below Can you help me on this issue. In one of the forum topics you were mentioning introduce country and area codes. I did that but it only changes dialed number.

 

I couldn't really understand the difference between sip registration and sip gateway. My sip provider can do the both which one should I use?

 

2] 2012/04/01 23:22:26: Trunk status mydivert (3) changed to "200 OK" (Refresh interval 1800 seconds)

[1] 2012/04/01 23:28:42: UDP: TOS could not be set

[1] 2012/04/01 23:28:43: Last message repeated 2 times

[5] 2012/04/01 23:28:43: Dialplan "Standard Dialplan": Match 00905322865600@192.168.66.41 to <sip:00905322865600@sip.mydivert.com;user=phone> on trunk mydivert

[1] 2012/04/01 23:28:43: UDP: TOS could not be set

[1] 2012/04/01 23:28:43: Last message repeated 2 times

[5] 2012/04/01 23:28:43: INVITE Response 603 Declined: Terminate adeadbdb@pbx

[1] 2012/04/01 23:41:53: UDP: TOS could not be set

[1] 2012/04/01 23:41:53: Last message repeated 2 times

[5] 2012/04/01 23:41:53: Dialplan "Standard Dialplan": Match 05322865600@192.168.66.41 to <sip:05322865600@sip.mydivert.com;user=phone> on trunk mydivert

[1] 2012/04/01 23:41:53: UDP: TOS could not be set

[1] 2012/04/01 23:41:53: Last message repeated 2 times

[5] 2012/04/01 23:41:53: INVITE Response 603 Declined: Terminate 2924b7d2@pbx

[1] 2012/04/01 23:42:20: UDP: TOS could not be set

[1] 2012/04/01 23:42:20: Last message repeated 2 times

[5] 2012/04/01 23:42:20: Dialplan "Standard Dialplan": Match 05322865600@192.168.66.41 to <sip:05322865600@sip.mydivert.com;user=phone> on trunk mydivert

[1] 2012/04/01 23:42:20: UDP: TOS could not be set

[1] 2012/04/01 23:42:20: Last message repeated 2 times

[5] 2012/04/01 23:42:20: INVITE Response 603 Declined: Terminate 4fa435b8@pbx

[1] 2012/04/01 23:43:33: UDP: TOS could not be set

[1] 2012/04/01 23:43:33: Last message repeated 2 times

[5] 2012/04/01 23:43:33: Dialplan "Standard Dialplan": Match 5808236@192.168.66.41 to <sip:5808236@sip.mydivert.com;user=phone> on trunk mydivert

[1] 2012/04/01 23:43:33: UDP: TOS could not be set

[1] 2012/04/01 23:43:34: Last message repeated 2 times

[5] 2012/04/01 23:43:34: INVITE Response 603 Declined: Terminate 25d87fbb@pbx

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To see why the provider returns 6xx codes, it would be useful to see the SIP INVITE packet that the PBX sends to the service provider. Version 4.5 now is very flexible with the SIP headers, so it should be possible to tweak the settings to make it happen.

 

SIP registration is useful when you operate the PBX on a privat (unroutable) IP address and you provider supports Session Border Controller functionality. Otherwise gateway is also fine, especially when your provider wants to send you the DID number and he does not support the P-Called-Party-ID header.

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İi tought that i managed to set the trunk properly since it was running ok last night. This morning i checked it out and seen my favorite messega 408 time out. From the logs it seems that it has a problem with dns server. It is somehow looking to 127,0,0.1 as dns server. It says that sip. Mydivert.com expired. I tried the ip address as server it is not working as well

 

I just seen a post relating to dns issue. Is there any solution or what ismy problem

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i just added google public dns and restarted the snomone once again. here is the log. it seems that it resolves the dns no problem but not registering

 

 

[7] 2012/04/02 08:50:06: syslog is disabled on the PBX

[1] 2012/04/02 08:50:06: Starting up version 2011-4.3.0.5021

[3] 2012/04/02 08:50:06: Call limit reached. Call ports 0, g729 0

[7] 2012/04/02 08:50:06: Found time zones HST AKDT AKST PDT PST MDT MST CDT CST2 EDT EST ADT AST NDT NST BST CET GMT+2 TUR GMT+3 GMT+4 GMT+6 GMT+7 GMT+8 GMT+9 GMT+10 GMT+11 GMT+12 CST CAT IST AUS1 AUS2 AUS3 AUS4 AUS5 AUS6 NZST CHAST GMT

[8] 2012/04/02 08:50:06: Scheduler precision is 0 us

[1] 2012/04/02 08:50:06: Working Directory is /Applications/snomONE/conf

[7] 2012/04/02 08:50:06: UDP: Opening socket on 0.0.0.0

[7] 2012/04/02 08:50:06: UDP: Opening socket on [::]

[8] 2012/04/02 08:50:06: No from address for sending text email

[5] 2012/04/02 08:50:06: Starting threads

[7] 2012/04/02 08:50:06: UDP: Opening socket on 0.0.0.0

[7] 2012/04/02 08:50:06: UDP: Opening socket on [::]

[7] 2012/04/02 08:50:06: UDP: Opening socket on 0.0.0.0:5060

[8] 2012/04/02 08:50:06: Joined multicast group 224.0.1.75

[7] 2012/04/02 08:50:06: UDP: Opening socket on [::]:5060

[1] 2012/04/02 08:50:06: UDP: TOS could not be set

[7] 2012/04/02 08:50:06: TCP: Opening socket on 0.0.0.0:5060

[7] 2012/04/02 08:50:06: TCP: Opening socket on [::]:5060

[7] 2012/04/02 08:50:06: TCP: Opening socket on 0.0.0.0:5061

[7] 2012/04/02 08:50:06: TCP: Opening socket on [::]:5061

[7] 2012/04/02 08:50:06: TCP: Opening socket on 0.0.0.0:81

[7] 2012/04/02 08:50:06: TCP: Opening socket on [::]:81

[7] 2012/04/02 08:50:06: TCP: Opening socket on 0.0.0.0:442

[7] 2012/04/02 08:50:06: TCP: Opening socket on [::]:442

[7] 2012/04/02 08:50:06: TCP: Opening socket on 0.0.0.0:388

[7] 2012/04/02 08:50:06: TCP: Opening socket on [::]:388

[7] 2012/04/02 08:50:06: UDP: Opening socket on 0.0.0.0:161

[7] 2012/04/02 08:50:06: UDP: Opening socket on [::]:161

[7] 2012/04/02 08:50:06: UDP: Opening socket on 0.0.0.0:69

[7] 2012/04/02 08:50:06: UDP: Opening socket on [::]:69

[8] 2012/04/02 08:50:06: Trunk 1: sending discover message for please.change

[5] 2012/04/02 08:50:06: Set scheduling priority to 15

[8] 2012/04/02 08:50:06: Trunk 1: Received reply for discover method

[8] 2012/04/02 08:50:06: Trunk 1 (PSTN Gateway) is associated with the following addresses: 1.2.3.4

[8] 2012/04/02 08:50:06: Trunk 3: sending discover message for Sip.mydivert.com

[8] 2012/04/02 08:50:06: DNS: Use DNS server 8.8.8.8

[8] 2012/04/02 08:50:06: DNS: Request sip.mydivert.com from server 8.8.8.8

[8] 2012/04/02 08:50:06: DNS: Add NAPTR sip.mydivert.com (ttl=60)

[8] 2012/04/02 08:50:06: DNS: Request _sips._tcp.sip.mydivert.com from server 8.8.8.8

[8] 2012/04/02 08:50:06: DNS: Add SRV _sips._tcp.sip.mydivert.com (ttl=60)

[8] 2012/04/02 08:50:06: DNS: Request _sip._tcp.sip.mydivert.com from server 8.8.8.8

[8] 2012/04/02 08:50:06: DNS: Add SRV _sip._tcp.sip.mydivert.com (ttl=60)

[8] 2012/04/02 08:50:06: DNS: Request _sip._udp.sip.mydivert.com from server 8.8.8.8

[8] 2012/04/02 08:50:06: DNS: Add SRV _sip._udp.sip.mydivert.com (ttl=60)

[8] 2012/04/02 08:50:06: DNS: Request sip.mydivert.com from server 8.8.8.8

[8] 2012/04/02 08:50:06: DNS: Add AAAA sip.mydivert.com (ttl=60)

[8] 2012/04/02 08:50:06: DNS: Request sip.mydivert.com from server 8.8.8.8

[8] 2012/04/02 08:50:06: DNS: Add A sip.mydivert.com 78.46.43.9 (ttl=466)

[8] 2012/04/02 08:50:06: Trunk 3: Received reply for discover method

[8] 2012/04/02 08:50:06: Trunk 3 (mydivert) is associated with the following addresses: udp:78.46.43.9:5060

[8] 2012/04/02 08:50:06: Trunk mydivert: Sending registration to Sip.mydivert.com

[8] 2012/04/02 08:50:06: SIP Tx udp:78.46.43.9:5060:

REGISTER sip:sip.mydivert.com SIP/2.0

Via: SIP/2.0/UDP 192.168.66.41:5060;branch=z9hG4bK-b233955f43da71a538aed416038800dc;rport

From: "mydivert" <sip:001101432@Sip.mydivert.com>;tag=393221856

To: "mydivert" <sip:001101432@Sip.mydivert.com>

Call-ID: 3hv6vva3@pbx

CSeq: 22565 REGISTER

Max-Forwards: 70

Contact: <sip:001101432@192.168.66.41:5060;transport=udp;line=eccbc87e>;+sip.instance="<urn:uuid:65143a6f-76ab-43d1-b7dc-55dcf378785b>"

User-Agent: snom-PBX/2011-4.3.0.5021

Supported: outbound

Expires: 3600

Content-Length: 0

 

[8] 2012/04/02 08:50:07: SIP Tr udp:78.46.43.9:5060:

REGISTER sip:sip.mydivert.com SIP/2.0

Via: SIP/2.0/UDP 192.168.66.41:5060;branch=z9hG4bK-b233955f43da71a538aed416038800dc;rport

From: "mydivert" <sip:001101432@Sip.mydivert.com>;tag=393221856

To: "mydivert" <sip:001101432@Sip.mydivert.com>

Call-ID: 3hv6vva3@pbx

CSeq: 22565 REGISTER

Max-Forwards: 70

Contact: <sip:001101432@192.168.66.41:5060;transport=udp;line=eccbc87e>;+sip.instance="<urn:uuid:65143a6f-76ab-43d1-b7dc-55dcf378785b>"

User-Agent: snom-PBX/2011-4.3.0.5021

Supported: outbound

Expires: 3600

Content-Length: 0

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Well, it seems there is nothing coming back... There is not much that the PBX can do. Maybe the service providers is down or the firewall blocks the traffic.

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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

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I think there is not a solution to my problems. I will give up mac os but will use the machine. I am wondering if I use vmware with linux or win to operate snomone. What is the most reliable and and widely used operating system for snomone. If the linux is prefered which linux is to use for.

 

I would very much appreciate if you could help to me.

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Not sure if this is a OS related issue. If PBX is getting 603 Declined, then someone on the trunk side rejecting it. Maybe thy did not like the number itself.

 

But if you want to try it on another OS, Linux is popular. Lot of folks are using CentOS-64 version.

 

We also have the virtual appliance for vmware. Check out http://snomone.wordpress.com/2012/02/21/snomone-virtual-appliance/

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