Help setting up incoming with Vitelity

[joshabts]

Hello all,

I am new to 3CX and am looking at using it for my small business. However I am having trouble setting up this server with Vitelity service. I have read of many people using 3CX with Vitelity but I cannot seem to be able to configure it to receive incoming calls.

I have setup the VoIP provider and DID number in 3CX and it reports that the line successfully registers. However when I try calling the number it rings busy and I get a failed call notification from Vitelity saying CHANUNAVAIL. When I contact them they are telling me this is because I am not registered with their servers. However 3CX still reports that I am. I have NAT enabled in Vitelity panel and have forwarded ports 5060, 5061, and 10000-20000 to my 3CX server. I have also setup a DNS name with my IP since I have a dynamic address.

What other settings did people with Vitelity have to do or can someone give me a quick step by step for setting up service with Vitelity?

Any help would be greatly appreciated and I can provide any other settings people need to know to help me.

Thank you!
Joshua Abts

PS I also forgot to mention that I can make outgoing calls via Vitelity just fine.

 

[5qg4]

Please enable the log level to "Verbose (only used for debugging purposes)". Restart all 3cx services or restart 3cx box. Then post the log here again. It maybe speed up to fix your issue.

Have you config the DID rules?

 

[joshabts]

17:33:10.000|.\CallMgr.cpp(75)|Log2|CallManager|CallMgr::Initialize:[CM100000] Start 3CX PhoneSystem Call Manager
17:33:10.000|.\CallMgr.cpp(76)|Log2|CallManager|CallMgr::Initialize:[CM100001] Version: 3.1.2434.0
17:33:10.296|.\CallMgr.cpp(114)|Log2|CallManager|CallMgr::Initialize:[CM100002] Default Local IP address: 172.168.0.2:5060
17:33:10.296|.\CallMgr.cpp(122)|Log2|CallManager|CallMgr::Initialize:[CM100003] ** Initializing SIP stack **
17:33:10.312|.\CM_Stack.cpp(28)|Log2|SipStack|CallMgr::Stack::Initialize:[CM106000] ** Adding transports **
17:33:10.343|.\DBAccessor.cpp(85)|Log2||DBA:[CM109000] ** Database connection Ok **
17:33:10.359|.\SLServer.cpp(30)|Log2||ListenConnect:[CM114000] SL: connected voippbx:0/MediaServer at [voippbx]/MediaServer
17:33:10.359|.\MSInterface.cpp(686)|Log2||MediaServerConnected:[CM112000] Media Server is connected
17:33:10.515|.\CallMgr.cpp(244)|Log2|DUM|CallMgr:umThread::thread:[CM100004] ** Enter DUM Thread **
17:33:10.515|.\Registrar.cpp(42)|Log2|Registrar|ServRegs::checkExpiration:[CM113000] Registration for sip:[email protected] has expired
17:33:10.968|.\SLServer.cpp(30)|Log2||ListenConnect:[CM114000] SL: connected voippbx:0/VoiceBoxManagerService at [voippbx]/VoiceBoxManagerService
17:33:11.125|.\CM_Stack.cpp(88)|Log2|SipStack|CallMgr::Stack::thread:[CM106001] ** Enter Stack Loop **
17:33:11.234|.\StunClient.cpp(67)|Log2|STUN|StunClient:rocess:[CM115001] Send initial STUN request to 80.239.235.209
17:33:11.359|.\StunClient.cpp(84)|Log2|STUN|StunClient:rocess:[CM115002] STUN resolved external IP=24.222.111.151:5060 by server 80.239.235.209
17:33:12.078|.\StunClient.cpp(84)|Log2|STUN|StunClient:rocess:[CM115002] STUN resolved external IP=24.222.111.151:5060 by server 64.69.76.23
17:33:12.109|.\ExtLine.cpp(325)|Log2|Endpoints|ExtLine::Register:[CM110001] Use External IP for device line registration DN='20000' device='Vitelity 877#'
17:33:12.109|.\ExtLine.cpp(366)|Log2|Endpoints|ExtLine::Register:[CM110004] Send registration for "8777654321"<sip:[email protected]>
17:33:12.140|.\ExtLine.cpp(325)|Log2|Endpoints|ExtLine::Register:[CM110001] Use External IP for device line registration DN='20000' device='Vitelity 877#'
17:33:12.140|.\ExtLine.cpp(366)|Log2|Endpoints|ExtLine::Register:[CM110004] Send registration for "8777654321"<sip:[email protected]>
17:33:12.281|.\Registrar.cpp(141)|Log2|Registrar|ServRegs:nAdd:[CM113002] Registered: Ext.100
17:33:12.468|.\Registrar.cpp(381)|Log2|Registrar|ClientRegs:nSuccess:[CM113005] Registration of sip:[email protected] is successful
17:33:12.578|.\Registrar.cpp(381)|Log2|Registrar|ClientRegs:nSuccess:[CM113005] Registration of sip:[email protected] is successful
17:33:14.906|.\SLServer.cpp(30)|Log2||ListenConnect:[CM114000] SL: connected voippbx:5483/IVRServer at [voippbx]/IVRServer
17:33:14.906|.\IVRInterface.cpp(249)|Log2||IVRConnected:[CM111000] IVR Server is connected
17:35:05.781|.\SLServer.cpp(30)|Log2||ListenConnect:[CM114000] SL: connected voippbx:0/PHPExtension_0 at [voippbx]/PHPExtension_0
17:40:25.796|.\ExtLine.cpp(325)|Log2|Endpoints|ExtLine::Register:[CM110001] Use External IP for device line registration DN='20000' device='Vitelity 877#'
17:40:25.796|.\ExtLine.cpp(366)|Log2|Endpoints|ExtLine::Register:[CM110004] Send registration for "8777654321"<sip:[email protected]>
17:40:26.218|.\Registrar.cpp(381)|Log2|Registrar|ClientRegs:nSuccess:[CM113005] Registration of sip:[email protected] is successful
18:19:16.328|.\Registrar.cpp(195)|Log2|Registrar|ServRegs:nRemove:[CM113003] Unregistered: Ext.200
18:19:52.046|.\Registrar.cpp(141)|Log2|Registrar|ServRegs:nAdd:[CM113002] Registered: Ext.200
18:30:18.031|.\ExtLine.cpp(325)|Log2|Endpoints|ExtLine::Register:[CM110001] Use External IP for device line registration DN='20000' device='Vitelity 877#'
18:30:18.031|.\ExtLine.cpp(366)|Log2|Endpoints|ExtLine::Register:[CM110004] Send registration for "8777654321"<sip:[email protected]>
18:30:18.578|.\Registrar.cpp(381)|Log2|Registrar|ClientRegs:nSuccess:[CM113005] Registration of sip:[email protected] is successful
19:22:33.281|.\Registrar.cpp(195)|Log2|Registrar|ServRegs:nRemove:[CM113003] Unregistered: Ext.200
21:54:35.609|.\Registrar.cpp(141)|Log2|Registrar|ServRegs:nAdd:[CM113002] Registered: Ext.200

That is my log output from recent times. Numbers have been changed for privacy concerns.

I have created the VoIP provider but even when I created a new DID number that used the provider I had created, it still gave the same result.

Thanks!

 

[5qg4]

Hi joshabts,

Referred to your log, there is not found any incoming call. Please issue a call to your 3CX box. Then capture the log which shows your problem.


The following log shows "NO DID rule for 558"

12:28:35.328 CallConf:nIncoming [CM003002] Call(29): Destination not available for call from sip:[email protected] to sip:[email protected]
12:28:35.328 CallConf::Rejected [CM103005] Call(29) is rejected: Destination does not exist, or is not registered
12:28:35.296 CallConf:nIncoming [CM103002] Call(29): Incoming call from 28735648 (Ln:10600@DIDGW01) to sip:558@192.168.1.252

The following log shows "DID rule for 135, and ring EXT 135"

12:27:40.328 StratLink:nHangUp [CM104001] Call(28 ): Ln:10600@DIDGW01 hung up call; cause: BYE; from IP:192.168.1.253
12:27:35.328 CallLegImpl:nConnected [CM103001] Call(28 ): Created audio channel for Ext.135 (10.1.3.106:6604) with Media Server (192.168.1.252:7102)
12:27:35.312 StratInOut:nConnected [CM104005] Call(28 ) : Setup completed for call from Ln:10600@DIDGW01 to Ext.135
12:27:35.312 CallLegImpl:nConnected [CM103001] Call(28 ) : Created audio channel for Ln:10600@DIDGW01 (192.168.1.253:30158) with Media Server (192.168.1.252:7100)
12:27:22.765 CallConf:nProvisional [CM103003] Call(28 ) : Ext.135 is ringing
12:27:22.421 CallConf:nIncoming [CM103002] Call(28 ): Incoming call from 28735648 (Ln:10600@DIDGW01) to sip:135@192.168.1.252

Regards,
5QG4

 

[joshabts]

This was a log with multiple calls made to the DID number throughout it. From what I can see and what I understand you to be saying, is that the calls aren't even reaching my PBX. Hence nothing in the log.

That said, I spoke with Vitelity and they keep telling me I am not properly registered because they do a sip debug peer 'myuser' and they are getting no IP found for peer. Which tells them my PBX is not registered as it should be.

I am not sure where to check on what exactly is sent through the registration or what ports possibly need to be open for the registration to successfully complete all the way. Is there a control port I am missing or need to check on? Is there any log file that I can check on for why registration may not complete properly even though 3CX says it does and shows the line as registered?

Any one else having/had this problem?

 

[Mirzab]

For port forwarding please read http://www.3cx.com/forums/viewtopic.php?t=1697&highlight=port+sticky

Some good info there:

List of default ports used by the 3cx phonesystem:
5060 (stanard sip port) TCP/UDP
5061 (secondary sip port) TCP/UDP
5480 (portgre) (only on local machine)
5481 (apache) (only on local machine)
5482 (media server) (only on local machine)
3478 (STUN Server) TCP/UDP
7000-7500 (Default port for internall calls) (only on local machine)
9000 - 9003 (ports for external calls through VOIP Gateway) UDP
10000-20000 (RTP ports) (Used if no stun available for audio)

Didn't see some of these (eg. 9000 - 9003) in your list posted. And even though this might seem self-evident, make sure forwarding is both ways (WAN <=> LAN IP) as needed.

Hope this helps.

Mirza.

 

[joshabts]

All of those ports have been opened and forwarded to my server. However I still have the same result when trying to receive calls.

I can though, make an outgoing call and it goes out through Vitelity just fine. This really sounds like a problem with their end does it not?

 

[Mirzab]

Just a suggestion since I have no knowledge or experience with that specific provider - check your current external IP and use that in their setup, just to test. Even a dynamic IP stays stable for a few hours. I have always tried to eliminate as many variables as possible when troubleshooting, and the domain forwarding for your dynamic IP is definitely an unknown.

I have also found it more stable in 3CX setting up my DID numbers as VoIP providers inbound only, works great for me whenever possible.

 

[5qg4]

Hi joshabts,

That said, I spoke with Vitelity and they keep telling me I am not properly registered because they do a sip debug peer 'myuser' and they are getting no IP found for peer. Which tells them my PBX is not registered as it should be.

Referred to your log, your WAN IP had been resolved by STUN. It highlighted in red color as below.

17:33:11.234|.\StunClient.cpp(67)|Log2|STUN|StunClient:rocess:[CM115001] Send initial STUN request to 80.239.235.209
17:33:11.359|.\StunClient.cpp(84)|Log2|STUN|StunClient:rocess:[CM115002] STUN resolved external IP=24.222.111.151:5060 by server 80.239.235.209
17:33:12.078|.\StunClient.cpp(84)|Log2|STUN|StunClient:rocess:[CM115002] STUN resolved external IP=24.222.111.151:5060 by server 64.69.76.23

Regards.

 

[tjabaut]

joshabts,

What are your registration settings for VoIP for Vitelity.

I mean could ytou please document out each and every option that you have enabled.

I have Vitelity and it has worked flawlessly for months now. I would be happy to help.

I know RobLloyd is also a fellow Vitelity user, and together we should be able to assit.

 

[joshabts]

Okay in this post I will try my best to cover all of the settings I currently have.

In Line management, I currently have one VoIP provider.

Provider Name: Vitelity 877#
External #: 8777654321
Authentication ID: myuser
Password: myvitelitysecret

Route Inbound calls during office hours to extension: 200
Route Inbound calls during non-office hours to extension: 200

Internal number: 20000
Outbound caller ID: 8777654321
Allow outbound calls: checked
Answer incoming calls: checked
Answer after: 0 seconds
Max simultaneous calls: 10

If I edit Providers/Gateways and edit my Vitelity 877# provider, these are those settings...

Registrar proxy hostname: inbound4.vitelity.net
Registrar proxy port: 5060
Outbound proxy hostname: outbound1.vitelity.net
Outbound proxy port: 5060
STUN Server hostname: stun.xten.com
STUN server port: 3478

Registration settings->
Time between registrations: 3600s
Require (optionally: authorized) registration for: Both in-out calls
Which IP to use in 'Contact' field for registration: External (STUN resolved)

Codec Priorities->
Available: Speex
Assigned codecs: G.711 U-law, G.711 A-law, GSM-FR

Provider capabilities->
Supports re-invite: unchecked
Supports replaces header: disabled

Location of destination number->
To.user field: checked
Request-Line-URI field: checked
'RemotePartyID;party=called' field: unchecked

Other options->
PBX Delivers audio: checked
Use IP in 'Contact' to recognize source: unchecked
IP Range: disabled
Caller-ID is in 'user' part of (one of following): From Field
Called-Number (for inbound calls) is in: To Field

Then under General Settings I have the following

SIP Port: 5060

Internal calls First Port: 7000
Internal calls Last Port: 7499

External calls first port: 9000
External calls last port: 9003

STUN Server options
Default STUN server address: stun.3cx.com
Default 2 STUN server: stun2.3cx.com
Timeout for STUN: 3000ms
Recheck every: 1200s

Settings for direct SIP calls
Allow calls to external URI: not checked
Local SIP domain: <blank>

Logging level: verbose



I think that pretty much covers all of the settings, if you need another page, let me know thanks!
Josh