Voicemail not working

[AHAutomation]

Hi,

Trying out 3cx on a VPS, everything works (as far as i know)

Voicemail is enabled within the settings of the extension.

When trying to call a phone and getting to voicemail (after 20 sec) i get the standard response "This call cannot be completed" also changing the "Voicemail language" is not working (changed it from english to dutch, also tried restarting the pbx)

Also tried to following with same result:
call *4405 (Where 405 = extension)
call 999 from extension 405


Activity Log:
31-Mar-2016 16:28:04.448 Leg L:7.2[VMail] is terminated: Cause: BYE from PBX
31-Mar-2016 16:28:04.448 [CM503008]: Call(C:7): Call is terminated
31-Mar-2016 16:28:04.437 Leg L:7.1[Extn] is terminated: Cause: BYE from 94.xxx.xxx.166:55665
31-Mar-2016 16:28:03.959 [MS205001] C:7.1 RTCP: Party address changed from 192.xxx.xxx.102:28412 to 94.xxx.xxx.166:11944
31-Mar-2016 16:28:03.864 [MS205001] C:7.1 RTP: Party address changed from 94.xxx.xxx.166:20330 to 94.xxx.xxx.166:7619
31-Mar-2016 16:27:59.313 [MS205001] C:7.1 RTP: Party address changed from 192.xxx.xxx.102:28411 to 94.xxx.xxx.166:20330
31-Mar-2016 16:27:58.833 [CM503007]: Call(C:7): VMail:999 has joined, contact <sip:[email protected]:5483>
31-Mar-2016 16:27:58.832 [CM503007]: Call(C:7): Extn:403 has joined, contact <sip:[email protected]:42716>
31-Mar-2016 16:27:58.829 L:7.2[VMail] has joined to L:7.1[Extn]
31-Mar-2016 16:27:58.829 NAT/ALG check:L:7.2[VMail] RESPONSE 200 on 'INVITE' - basic check passed. No information for extended checks
31-Mar-2016 16:27:58.681 [CM503025]: Call(C:7): Calling T:VMail:999@[Dev:sip:[email protected]:5483;rinstance=c9d2a40a953aca60] for L:7.1[Extn]
31-Mar-2016 16:27:58.628 [CM503027]: Call(C:7): From: Extn:403 ("403 - Peter" <sip:[email protected]:5060>) to T:VMail:999@[Dev:sip:[email protected]:5483;rinstance=c9d2a40a953aca60]
31-Mar-2016 16:27:58.628 [CM503004]: Call(C:7): Route 1: from L:7.1[Extn] to T:VMail:999@[Dev:sip:[email protected]:5483;rinstance=c9d2a40a953aca60]
31-Mar-2016 16:27:58.625 [CM503001]: Call(C:7): Incoming call from Extn:403 to <sip:*[email protected]:5060>
31-Mar-2016 16:27:58.624 NAT/ALG check:L:7.1[Extn] REQUEST 'INVITE' - some of SIP/SDP headers may contain inconsistent information or modified by intermediate hop
SIP contact header is not equal to the SIP packet source(IPort):
Contact address:192.xxx.xxx.102:42716
Received from :94.xxx.xxx.166:55665
Media session IP ('c=' attribute) is not equal to the SIP packet source(IPort):
Media session IP: 192.xxx.xxx.102
Received from: 94.xxx.xxx.166

Someone an idea?

 

[leejor]

Was this ever working?
You may have to explain more about your set-up. I suspect that the problems have something to do with ...

31-Mar-2016 16:27:58.624 NAT/ALG check:L:7.1[Extn] REQUEST 'INVITE' - some of SIP/SDP headers may contain inconsistent information or modified by intermediate hop
SIP contact header is not equal to the SIP packet source(IPort):

and...

31-Mar-2016 16:28:03.959 [MS205001] C:7.1 RTCP: Party address changed from 192.xxx.xxx.102:28412 to 94.xxx.xxx.166:11944
31-Mar-2016 16:28:03.864 [MS205001] C:7.1 RTP: Party address changed from 94.xxx.xxx.166:20330 to 94.xxx.xxx.166:7619
31-Mar-2016 16:27:59.313 [MS205001] C:7.1 RTP: Party address changed from 192.xxx.xxx.102:28411 to 94.xxx.xxx.166:20330

 

[NickD_3CX]

Just 2 questions:
- On the server where 3CX is installed, do you have a Public IP on the NIC?
- Did you choose Abyss or IIS as the Web Server during installation?

 

[AHAutomation]

@ Leejor.

I guess it never did..... never listened to the rest of what the voice had to say...
U address me some lines i have given. I can see the protocol is bouncing some RTP ports but making a call is no issue, the 3cx is also replying with a voice.

@ NickD_3CX.
My 3cx server is behind an PFSense. Outbound NAT ports are static.
I have chosen IIS, but i have changed the HTTP port, did some changes in parameters for sending call reports.

I also tested with 3CXPhone to call to *4405 got the same result.

Are there other places to look for error logs?

 

[AHAutomation]

Sorry Guys, i fixed it.

Last month had to change my remote Ports of HTTP and HTTPS.
I only changed these for management in parameters, so mails where send with the right PDF link for Call Reports.
MANAGEMENT_LINK_EXT
MANAGEMENT_LINK_EXT_SEC
MANAGEMENT_LINK_LOCAL
MANAGEMENT_LINK_LOCAL_SEC

I found some postings on 3cx forums, from upgrade errors from 11->12 that where caused by access ports.

So today I tested first to change the following HTTP and HTTPS ports:
IVR_STARTURL

After this my call was forwarded to the voicemail, only did end up at the voicemail, just ringed.

Than i changed all the parameters with an HTTP and HTTPS accesss.
CALLUS_LINK_EXT
CALLUS_LINK_EXT_SEC
CALLUS_LINK_LOCAL
CALLUS_LINK_LOCAL_SEC
MYPHONE_LINK_EXT
MYPHONE_LINK_EXT_SEC
MYPHONE_LINK_LOCAL
MYPHONE_LINK_LOCAL_SEC
PROVISIONING_LINK_EXT
PROVISIONING_LINK_EXT_SEC
PROVISIONING_LINK_LOCAL
PROVISIONING_LINK_LOCAL_SEC
REPORTER_LINK_EXT
REPORTER_LINK_EXT_SEC
REPORTER_LINK_LOCAL
REPORTER_LINK_LOCAL_SEC
WEB_ROOT_EXT
WEB_ROOT_EXT_SEC
WEB_ROOT_LOCAL
WEB_ROOT_LOCAL_SEC

After this my voicemail worked!!

 

[NickD_3CX]

@ NickD_3CX.
My 3cx server is behind an PFSense. Outbound NAT ports are static.
I have chosen IIS, but i have changed the HTTP port, did some changes in parameters for sending call reports.

I still didn't understand of the NIC of the 3CX Server has a Private IP or a Public IP.

Anyhow, if you have changed the ports that well may be the issue. The reason is that the 3CX SIP Server sends internal HTTP/S requests to the IVR Server in order to start to the default HTTP/S ports. The IVRs that will be affected include the VMail, Conference IVR, etc. Also, the HTTP ports to which the SIP Server makes requests cannot be changed as far as I am aware.

Try changing the ports back to what they were and try again, at least to see if this is the issue.