New to 3CX - need some help on system design

[h2009]

Hi there,

I've just looked into 3CX and it seems to do everything i need. At the moment im just trying to put all the pieces of the puzzel together and could use with some help.

I'll be running 3CX from a shuttle PC - which will always be on, and only used for that.
The phones i'll be using are the linksys SPA921 models - 4 will be connected via ethernet, 6 via wireless pack (linksys WBP54g).

Im trying to also have a backup PSTN system should my broadband connection fail. And for that im thinking about using the linksys SPA3102.

As far as i know of the handsets and the analoge adaptor should work fine with 3CX (as its said on the webpage) - i'd be greatful if someone could confirm this to me.

But the problem now is finding a VOIP provider that can meet my needs and work with 3CX!

I live in the UK - and most of my calls will be made to the UK (landline and mobile), Dubai (landline and mobile), Iraq (landline). I need to keep my existing UK landline number from BT. At the moment the only provider i can find is Vonage, but i dont know it it will work with everything above, and the call cost is expensive on them.
Does anyone know of any other provider which can work, and provide me with call lower costs?

Also if this setup is successful, i need to roll out another UK VOIP system and a VOIP in Dubai - what would be the best way to go about this - i.e. free calls between each other?


--------------------------

Edit:

My existing network setup is :
Cable modem > wireless router > Hub > Wireless AP & ethernet points

Planned setup after:
Cable modem > wireless router > Hub & SPA3102 > Wireless AP & ethernet points & telephones

Will this work ok?


thanks for the help.

 

[Halea]

I've been playing with 3CX for a little over 4 months now. I have tested three different VOIP providers, one from the US and two from Europe - I have six such lines active. On my setup, I also have two PSTN lines from two different local phone companies; one terminated on SPA-3102 and the other on HT-488. My US based VOIP provider, InPhonex (actually a Canadian company with local presence) has been extremely easy to setup, very dependable, and very consistent. The other VIOP providers were average to good. SPA-3102 was very easy to setup, while HT488 was a nightmare, although eventually I got everything to work. Why am I telling you this? Just to prepare you for what's waiting around the corner. Some things will be very easy, some things a bit more complicated, eventually almost everything will work but you will need more and there will be always this little something that won't work exactly the way you wanted - or won't work at all.

To answer your questions; yes your requirements are rather basic. The VOIP provider choice is critical, but always remember that you are not geographically limited. Sure, you will hear that you will have more optimized timings with local providers rather than with one half way around the globe, but that's not always true. I'm dealing with a European VOIP which gives me excellent bandwidth and latency figures. Any decent VOIP will work with 3CX. That's one area where I personally didn't run into any significant problems (beside my 3 providers, I also tried more than a dozen freebies - they all worked pretty well).

It sounds like you already made a good choice for your gateway device. Go for a SPA-3102, NEVER CONSIDER HT488, so that you never have to look back! Of course many will disagree and tell you that HT488 works equally well. But why to spend many hours to figure out something on that device when you can get the same thing accomplished in 10 minutes with the first one?

Regarding your Dubai office, if you need simply a few extensions, just configure remote extensions to your UK system. I've done that without any problems. The question is; are you ok that a locall call in Dubai will have to be dialed from UK? The answer depends upon the rates, your business, accounting, etc. Another option is to run a soft tunnel between Dubai and UK where your main PBX is, but that doesn't address the local call issue. Finally, if you are going to have a local provider in Dubai, then you might want to install a second 3CX PBX over there (watch the extra expense, you will be paying for two licenses) and get your two PBXs bridged. There will be no expenses calling intra-company between Dubai and UK in any of the above configurations.

Linksys SPA921 works just fine with 3CX although my preference goes for GXP-2000. I simply like that GXP-2000 gives you 4 totally independent lines that you can configure any which way you like. And it's a few bucks less expensive (at least in the US)

The network side of the story will play out when you run your tests. Impossible to say what's going to happen, but most likely things will work just fine. The critical things are speed and latency of the cable line; open or closed tcp/udp ports per your internet provider's "safe" setup; your own gateway's settings, your own router's settings, the protections on the computer where you will be running 3CX and where you might have other software running a webserver, PHP, a database, etc. This is why I would sugest that you start with the free edition and if everything works to your satisfaction, then move up to your more elaborate and paid configuration.

Hope this helps a bit. Good luck. Welcome to the wonderful world of VOIP.
Halea

 

[h2009]

Hi Helea,

Thanks for your reply - Its been very helpful and atleast now i can do what needs to be done!

I've started setting up 3CX, and i can call into the network, but im having problems making calls out. I cant seem to find out why, and the logs dont really help at all.
Surely if i can call into the network i should be able to call?

At the moment im using Sipgate, but i just cant find any reason why outbound calls are not working.

 

[Halea]

I've had a sipgate (UK) account that I plugged back in 3CX. It worked flowlessly. I don't have any credits on it so I could only dial out their 10000 test line, and I could call in onto my UK phone number. Everything was fine in both directions.
Do not worry about the STUN server thing, you should not need it.
Can you tell me more about your outdial error condition? What do you hear in the handset as things fail?
Halea

 

[h2009]

ok here is the log from 3CX - im looking at my firewall logs too, there blocking some outbound traffic (even thou the 3CX is in the DMZ)

02:53:46.265 Call::Terminate [CM503008]: Call(1): Call is terminated
02:53:46.265 MediaServerReporting::SetRemoteParty [MS210000] C:1.1:Offer received. RTP connection: 192.168.0.200:8000(8001)
02:53:46.265 Call::RouteFailed [CM503014]: Call(1): Attempt to reach [sip:[email protected]] failed. Reason: Not Found
02:53:46.265 CallCtrl:nSelectRouteReq [CM503013]: Call(1): No known route to target: [sip:[email protected]]
02:53:46.250 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
02:53:46.250 CallLeg::setRemoteSdp Remote SDP is set for legC:1.1
02:53:46.250 Extension:rintEndpointInfo [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [NCH Swift Sound Express Talk 3.04] Transport: [sip:192.168.0.200:5060]
02:53:46.218 CallCtrl:nIncomingCall [CM503001]: Call(1): Incoming call from Ext.101 to [sip:[email protected]]
02:53:46.218 CallLeg:nNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.196:5060;rport=5060;branch=z9hG4bK80255
Max-Forwards: 20
Contact: [sip:[email protected]:5060]
To: [sip:[email protected]]
From: [sip:[email protected]];tag=9253
Call-ID: 1205715515-255-Layth%20Hamza%E2%80%99s%20MacBook%[email protected]
CSeq: 943 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
Proxy-Authorization: Digest username="101",realm="3CXPhoneSystem",nonce="12850196025:20fc37336986aaa8ea66161f572f44fc",uri="sip:[email protected]",response="2c642edc44fd9d152f8f4bf8c68c6e74",opaque="",algorithm=MD5
Supported: replaces
User-Agent: NCH Swift Sound Express Talk 3.04
Content-Length: 0


02:53:42.281 ClientRegs:nSuccess [CM504004]: Registration succeeded for: 1@Sipgate
02:53:41.921 ClientRegs::Register [CM504003]: Sent registration request for 1@Sipgate
02:53:40.625 ClientRegs::Register [CM504003]: Sent registration request for 1@Sipgate
02:53:40.312 ExtnCfg::updateContact [CM504001]: Ext.100: new contact is registered. Contact(s): [sip:[email protected]:5070;rinstance=1e08fb4b63455b07/100]
02:53:40.000 ExtnCfg::updateContact [CM504001]: Ext.101: new contact is registered. Contact(s): [sip:[email protected]:5060/101]

 

[h2009]

Here is a log of an internal call sucessfully ringing:

02:57:02.093 Call::Terminate [CM503008]: Call(2): Call is terminated
02:57:01.484 Extension:rintEndpointInfo [CM505001]: Ext.100: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX VoIP Client;Rev: 1] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Phone v0.1] Transport: [sip:192.168.0.200:5060]
02:57:01.484 CallCtrl:nAnsweredCall [CM503002]: Call(2): Alerting sip:[email protected]:5070;rinstance=1e08fb4b63455b07
02:57:00.453 MediaServerReporting::SetRemoteParty [MS210006] C:2.2:Offer provided. Connection(by pass mode): 192.168.0.200:8000(8001)
02:57:00.437 MediaServerReporting::SetRemoteParty [MS210000] C:2.1:Offer received. RTP connection: 192.168.0.200:8000(8001)
02:57:00.437 CallCtrl:nSelectRouteReq [CM503004]: Call(2): Calling: Ext:100@[Dev:sip:[email protected]:5070;rinstance=1e08fb4b63455b07]
02:57:00.437 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
02:57:00.437 CallLeg::setRemoteSdp Remote SDP is set for legC:2.1
02:57:00.437 Extension:rintEndpointInfo [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [NCH Swift Sound Express Talk 3.04] Transport: [sip:192.168.0.200:5060]
02:57:00.421 CallCtrl:nIncomingCall [CM503001]: Call(2): Incoming call from Ext.101 to [sip:[email protected]]
02:57:00.421 CallLeg:nNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.196:5060;rport=5060;branch=z9hG4bK82255
Max-Forwards: 20
Contact: [sip:[email protected]:5060]
To: [sip:[email protected]]
From: [sip:[email protected]];tag=9254
Call-ID: 1205715516-255-Layth%20Hamza%E2%80%99s%20MacBook%[email protected]
CSeq: 850 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
Proxy-Authorization: Digest username="101",realm="3CXPhoneSystem",nonce="12850196219:7d6fc99deffe6ee278a321a900df8877",uri="sip:[email protected]",response="11c0747ff56ba558768546b713063b64",opaque="",algorithm=MD5
Supported: replaces
User-Agent: NCH Swift Sound Express Talk 3.04
Content-Length: 0

 

[h2009]

Here is a log of outside call incoming:

02:58:34.046 Call::Terminate [CM503008]: Call(3): Call is terminated
02:58:34.031 LineCfg::getInboundTarget [CM503011]: Inbound out-of-office hours' rule for LN:1 forwards to DN:800
02:58:32.468 CallCtrl:nAnsweredCall [CM503002]: Call(3): Alerting sip:[email protected]:5070;rinstance=1e08fb4b63455b07
02:58:32.343 CallLeg:nFailure [CM503003]: Call(3): Call to sip:[email protected] has failed; Cause: 404 User unknown.; from IP:192.168.0.200:5060
02:58:32.234 AuthMgr:nAuthFailure [CM102001]: Authentication failed for SipReq: INVITE [email protected]:5060 tid=c133b572c172e835 cseq=INVITE [email protected]:5060 / 2 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings
02:58:32.218 evt::CheckAuth::not_handled [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-d87543-c133b572c172e835-1--d87543-;rport=5060
Max-Forwards: 70
Contact: [sip:[email protected]:5060]
To: [sip:[email protected]]
From: "07859932639"[sip:[email protected]:5060];tag=f658ce5d
Call-ID: NDVkZDJhMDE4M2FlNGZlYTJiMmMwMjEwYzExNzM5NWM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
Proxy-Authorization: Digest username="",realm="3CXPhoneSystem",nonce="12850196312:e172c89244a7bcdc8a0b0867e56f3207",uri="sip:[email protected]:5060",response="515664a5a99f9077eb0e7ddb6e6a5a5c",algorithm=MD5
User-Agent: 3CXPhoneSystem 5.1.4128.0
Content-Length: 0


02:58:32.218 evt::CheckAuth::not_handled [CM302002]: Authentication failed due to unidentified source of: SipReq: INVITE [email protected]:5060 tid=c133b572c172e835 cseq=INVITE [email protected]:5060 / 2 from(wire)
02:58:32.000 evt::CheckIfAuthIsRequired::not_handled [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-d87543-31499958f27a8f66-1--d87543-;rport=5060
Max-Forwards: 70
Contact: [sip:[email protected]:5060]
To: [sip:[email protected]]
From: "07859932639"[sip:[email protected]:5060];tag=f658ce5d
Call-ID: NDVkZDJhMDE4M2FlNGZlYTJiMmMwMjEwYzExNzM5NWM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
User-Agent: 3CXPhoneSystem 5.1.4128.0
Content-Length: 0


02:58:32.000 evt::CheckIfAuthIsRequired::not_handled [CM302001]: Authorization system can not identify source of: SipReq: INVITE [email protected]:5060 tid=31499958f27a8f66 cseq=INVITE [email protected]:5060 / 1 from(wire)
02:58:31.921 MediaServerReporting::SetRemoteParty [MS210002] C:3.4:Offer provided. Connection(transcoding mode): 127.0.0.1:7062(7063)
02:58:31.906 MediaServerReporting::SetRemoteParty [MS210002] C:3.3:Offer provided. Connection(transcoding mode): 192.168.0.200:7060(7061)
02:58:31.906 MediaServerReporting::SetRemoteParty [MS210002] C:3.2:Offer provided. Connection(transcoding mode): 192.168.0.200:7058(7059)
02:58:31.890 CallCtrl:nSelectRouteReq [CM503004]: Call(3): Calling: RingAll:800@[Dev:sip:[email protected]:5070;rinstance=1e08fb4b63455b07, Dev:sip:[email protected]:5060, Dev:sip:[email protected]:5060]
02:58:31.875 MediaServerReporting::SetRemoteParty [MS210000] C:3.1:Offer received. RTP connection: 217.10.68.74:36050(36051)
02:58:31.859 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]:5060]
02:58:31.859 CallLeg::setRemoteSdp Remote SDP is set for legC:3.1
02:58:31.859 Line:rintEndpointInfo [CM505003]: Provider:[Sipgate] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.0.200:5060]
02:58:31.859 LineCfg::getInboundTarget [CM503011]: Inbound out-of-office hours' rule for LN:1 forwards to DN:800
02:58:31.843 CallCtrl:nIncomingCall [CM503001]: Call(3): Incoming call from 07859932639@(Ln.1@Sipgate) to [sip:[email protected]:5060]
02:58:31.765 LineCfg::getInboundTarget [CM503011]: Inbound out-of-office hours' rule for LN:1 forwards to DN:800
02:58:31.734 CallLeg:nNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060;rinstance=560599671872ff7e SIP/2.0
Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bKb5cb.70847625.0
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bKb5cb.70847625.0
Via: SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.6;branch=z9hG4bK50960c6c
Via: SIP/2.0/UDP 217.10.69.13:5060;branch=z9hG4bK50960c6c;rport=5060
Max-Forwards: 67
Record-Route: [sip:217.10.79.23;lr=on;ftag=as16846d83]
Record-Route: [sip:172.20.40.2;lr=on]
Record-Route: [sip:217.10.79.23;lr=on;ftag=as16846d83]
Contact: [sip:[email protected]]
To: [sip:[email protected]]
From: "07859932639"[sip:[email protected]];tag=as16846d83
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

 

[Halea]

In your VOIP provider setup for SipGate (in 3CX), what do you have for outbound proxy? If there is anything, remove it and leave it blank. Give a try and let me know.
Halea

 

[h2009]

Im afraid that its already blank.

 

[Halea]

The following is a successful call from my 3CX extension 124 to phone number 10000 at SipGate. You can see that SipGate is mapped as 3CX external line #10010. My 3CX server is at IP 192.168.1.1. My phone at extension 124 is GXP-2000 at IP 192.168.1.2.
This is a medium detail level log. As you can see at one point you have:
Calling: VoIPline:10010@[Dev:sip:[email protected]:5060], then:
Alerting sip:[email protected]:5060, and later you have:
Device joined: sip:[email protected]:5060
I can't see something similar in your logs!
Halea

----- Below successful call to 10000 at SipGate ----
23:27:33.000 Call::Terminate [CM503008]: Call(29): Call is terminated
23:27:32.870 Call::Terminate [CM503008]: Call(29): Call is terminated
23:27:28.374 CallCtrl:nLegConnected [CM503007]: Call(29): Device joined: sip:[email protected]:5060
23:27:28.314 CallCtrl:nLegConnected [CM503007]: Call(29): Device joined: sip:[email protected]:5066;transport=udp
23:27:28.304 Line:rintEndpointInfo [CM505003]: Provider:[SipGate_UK] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.1.1:5060]
23:27:28.304 CallCtrl:nAnsweredCall [CM503002]: Call(29): Alerting sip:[email protected]:5060
23:27:27.593 CallCtrl:nSelectRouteReq [CM503004]: Call(29): Calling: VoIPline:10010@[Dev:sip:[email protected]:5060]
23:27:27.542 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
23:27:27.542 Extension:rintEndpointInfo [CM505001]: Ext.124: Device info: Device Identified: [Man: GrandStream;Mod: GXP-2000;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [Grandstream GXP2000 1.1.5.15] Transport: [sip:192.168.1.1:5060]
23:27:27.462 CallCtrl:nIncomingCall [CM503001]: Call(29): Incoming call from Ext.124 to [sip:[email protected]]

 

[h2009]

Ok here we go not there yet but a little improvement:

03:44:15.843 Call::Terminate [CM503008]: Call(12): Call is terminated
03:44:15.828 Call::RouteFailed [CM503014]: Call(12): Attempt to reach [sip:[email protected]] failed. Reason: Forbidden
03:44:15.828 CallLeg:nFailure [CM503003]: Call(12): Call to sip:[email protected]:5060 has failed; Cause: 603 Declined; from IP:217.10.79.23:5060
03:44:12.812 Line:rintEndpointInfo [CM505003]: Provider:[Sipgate] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.0.200:5060]
03:44:12.812 CallCtrl:nAnsweredCall [CM503002]: Call(12): Alerting sip:[email protected]:5060
03:44:12.218 MediaServerReporting::SetRemoteParty [MS210002] C:12.2:Offer provided. Connection(transcoding mode): 81.98.161.25:9010(9011)
03:44:12.140 MediaServerReporting::SetRemoteParty [MS210000] C:12.1:Offer received. RTP connection: 192.168.0.200:8000(8001)
03:44:12.140 CallCtrl:nSelectRouteReq [CM503004]: Call(12): Calling: VoIPline:1@[Dev:sip:[email protected]:5060]
03:44:12.140 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
03:44:12.140 CallLeg::setRemoteSdp Remote SDP is set for legC:12.1
03:44:12.140 Extension:rintEndpointInfo [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [NCH Swift Sound Express Talk 3.04] Transport: [sip:192.168.0.200:5060]
03:44:12.125 CallCtrl:nIncomingCall [CM503001]: Call(12): Incoming call from Ext.101 to [sip:[email protected]]
03:44:12.125 CallLeg:nNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.196:5060;rport=5060;branch=z9hG4bK103255
Max-Forwards: 20
Contact: [sip:[email protected]:5060]
To: [sip:[email protected]]
From: [sip:[email protected]];tag=9264
Call-ID: 1205715525-255-Layth%20Hamza%E2%80%99s%20MacBook%[email protected]
CSeq: 663 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
Proxy-Authorization: Digest username="101",realm="3CXPhoneSystem",nonce="12850199051:c545bf248901ff39df26a4ee05a9a6f2",uri="sip:[email protected]",response="04be64ddc614b9ea7c5fdf0b6937b2fb",opaque="",algorithm=MD5
Supported: replaces
User-Agent: NCH Swift Sound Express Talk 3.04
Content-Length: 0

 

[h2009]

Here is a medium log for easy viewing:

Server status
HOME | LOGOUT
3CX Phone System Version5.1.4128.0 PBX FAX DB DR MS VM TNL WEB


The server log shows server activity and detailed logging Messages. More > These log messages can allow you to troubleshoot firewall, VOIP gateway, VOIP provider and SIP phone configuration issues. When requesting support, always post the relevant server status logs for a particular issue. < Less

Time Function Message
03:45:31.828 Call::Terminate [CM503008]: Call(1): Call is terminated
03:45:31.812 Call::RouteFailed [CM503014]: Call(1): Attempt to reach [sip:[email protected]] failed. Reason: Forbidden
03:45:31.812 CallLeg:nFailure [CM503003]: Call(1): Call to sip:[email protected]:5060 has failed; Cause: 603 Declined; from IP:217.10.79.23:5060
03:45:28.796 Line:rintEndpointInfo [CM505003]: Provider:[Sipgate] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.0.200:5060]
03:45:28.796 CallCtrl:nAnsweredCall [CM503002]: Call(1): Alerting sip:[email protected]:5060
03:45:28.093 CallCtrl:nSelectRouteReq [CM503004]: Call(1): Calling: VoIPline:1@[Dev:sip:[email protected]:5060]
03:45:28.078 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
03:45:28.078 Extension:rintEndpointInfo [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [NCH Swift Sound Express Talk 3.04] Transport: [sip:192.168.0.200:5060]
03:45:28.046 CallCtrl:nIncomingCall [CM503001]: Call(1): Incoming call from Ext.101 to [sip:[email protected]]

 

[Halea]

Ok let's focus on the following (from your logs):
03:44:15.828 Call::RouteFailed [CM503014]: Call(12): Attempt to reach [sip:[email protected]] failed. Reason: Forbidden
03:44:15.828 CallLeg:nFailure [CM503003]: Call(12): Call to sip:[email protected]:5060 has failed; Cause: 603 Declined; from IP:217.10.79.23:5060

Try to reformulate the phone number as 10000 rather than 010000.
Also, any idea what's the IP{ 217.10.79.23? Is it the resolution of sipgate.co.uk? (actually i can check it myself :lol: )
Halea

 

[Halea]

Do you have a 0 prefix that you are forgetting to strip in your dial out rules?
Halea

 

[h2009]

Im just trying to figure out why i need that '0' there. At the moment im adjust the outbound calls, and putting a prepend - does that just cover up the problem?

At the moment these are my settings in outbound (when the log was taken):

Prefix 0
call from extension : blank
Calls to number with lenght : blank

route 1 : sipgate, strip digit 0, prepend 0.


FYI: when i dial from the phone im manually input the number: 010000,

If i put in 10000, the log looks like this:

03:58:10.546 Call::Terminate [CM503008]: Call(2): Call is terminated
03:58:10.531 Call::RouteFailed [CM503014]: Call(2): Attempt to reach [sip:[email protected]] failed. Reason: Not Found
03:58:10.531 CallCtrl:nSelectRouteReq [CM503013]: Call(2): No known route to target: [sip:[email protected]]
03:58:10.531 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
03:58:10.531 Extension:rintEndpointInfo [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [NCH Swift Sound Express Talk 3.04] Transport: [sip:192.168.0.200:5060]
03:58:10.515 CallCtrl:nIncomingCall [CM503001]: Call(2): Incoming call from Ext.101 to [sip:[email protected]]

 

[h2009]

If i put a strip digit of 1, and dial 010000, this is what i get:

04:00:14.031 Call::Terminate [CM503008]: Call(2): Call is terminated
04:00:14.031 Call::Terminate [CM503008]: Call(2): Call is terminated
03:59:54.750 CallCtrl:nLegConnected [CM503007]: Call(2): Device joined: sip:[email protected]:5060
03:59:54.734 CallCtrl:nLegConnected [CM503007]: Call(2): Device joined: sip:[email protected]:5060
03:59:54.734 Line:rintEndpointInfo [CM505003]: Provider:[Sipgate] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] Transport: [sip:192.168.0.200:5060]
03:59:54.734 CallCtrl:nAnsweredCall [CM503002]: Call(2): Alerting sip:[email protected]:5060
03:59:54.265 CallCtrl:nSelectRouteReq [CM503004]: Call(2): Calling: VoIPline:1@[Dev:sip:[email protected]:5060]
03:59:54.250 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
03:59:54.250 Extension:rintEndpointInfo [CM505001]: Ext.101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [NCH Swift Sound Express Talk 3.04] Transport: [sip:192.168.0.200:5060]
03:59:54.250 CallCtrl:nIncomingCall [CM503001]: Call(2): Incoming call from Ext.101 to [sip:[email protected]

 

[Halea]

My suggestion to you is to not use a number prefix (like 0) in your dial out rule for the time being. You can still have an outbound rule, but simply identify the source extension from which you are calling (like 101), and do not strip or append anything. You should be able to call 10000. Actually if you called 010000, you should get a ring and then an error message accompanied by an error ring tone (I've just tried it).
Halea

 

[h2009]

How would i go about it so that i would only have to dial 10000?
Im trying to do it now, but i cant seem to get it.

 

[Halea]

I think you're on the right track. The only thing that doesn't look good is the phone number that you're dialing.
Halea

 

[h2009]

Oh by the way isnt there meant to be a recorded message on the end of line? Because i can hear it ring, but soon as it answers theres nothing there?