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Voicemail is silent

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zeki

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I am testing the free version and can't access voicemail. The 3cx call assistant shows that I have two voicemails waiting, but when I phone the voicemail no (999), the call is answered and then I hear nothing. Voicemail is enabed for the extn. and 999 is the right no. Any ideas why i am not being asked for extn no and pin?
 
What does the log file show for the time period you try accessing voice mail? Make sure you have the log set to debug so you can get an idea of where to start looking.
 
This time i tried to enter the extension number even though i couldnt hear anything - the system didn't ask for it. Here is the log but I can't see a debug option anywhere.

Time Function Message
07:39:56.870 StratInOut::eek:nCancel [CM104009] Call(31): Call from Ext.101 to 999 has been terminated
07:39:54.434 MediaServerReporting::DTMFhandler [MS211000] Call(31) Ext.101: DTMF (RTP) from 127.0.0.1:63502 arrived. in-band DTMF tone detection is turned off.
07:39:41.425 MediaServerReporting::RTPSender [MS004000] Call(31) Ext.101: (ACTIVE): Can't send RTP stream to 127.0.0.1:63502 destination unreachable
07:39:41.285 CallLegImpl::eek:nConnected [CM103001] Call(31): Created audio channel for Ext.101 (127.0.0.1:63502) with Media Server (192.168.1.7:7060)
07:39:41.243 CallConf::eek:nIncoming [CM103002] Call(31): Incoming call from 101 (Ext.101) to sip:999@localhost
 
For what you have there, extension 101 isnt sending the right ip address to the server, thats the reason why its not working. if external try to bind that extensiion to media server and also check it has external....

Cheers
 
Sorry, but I'm a bit lost by your comments Miraportuga. The extension is not external. How do I bins to media server?
 
zeki said:
This time i tried to enter the extension number even though i couldnt hear anything - the system didn't ask for it. Here is the log but I can't see a debug option anywhere.

Time Function Message
07:39:56.870 StratInOut::eek:nCancel [CM104009] Call(31): Call from Ext.101 to 999 has been terminated
07:39:54.434 MediaServerReporting::DTMFhandler [MS211000] Call(31) Ext.101: DTMF (RTP) from 127.0.0.1:63502 arrived. in-band DTMF tone detection is turned off.
07:39:41.425 MediaServerReporting::RTPSender [MS004000] Call(31) Ext.101: (ACTIVE): Can't send RTP stream to 127.0.0.1:63502 destination unreachable
07:39:41.285 CallLegImpl::eek:nConnected [CM103001] Call(31): Created audio channel for Ext.101 (127.0.0.1:63502) with Media Server (192.168.1.7:7060)
07:39:41.243 CallConf::eek:nIncoming [CM103002] Call(31): Incoming call from 101 (Ext.101) to sip:999@localhost

Please specify which softphone was used to place a call.
 
I used x-lite - latest version
 
The softphone is reporting its IP as 127.0.0.1 which is invalid so connection is not completed properly. Does the PC with the softphone have multiple NICs?
 
yes, but one of them is disabled. I can make external calls using the softphone successfully its just the voicemail where i hear nothing!
 
Check logs after a successful external call with softphone - I bet Ext 101 shows a proper internal IP, not 127.0.0.1. Also try calling another extension (not the voicemail) and see which IP is reported in logs. If you don't have another extension install X-Lite on the 3CX machine and create an extension for it. Doesn't matter if you can't answer it, just the logs will narrow it if the invalid IP is specific to dialing 999.
 
x-lite is currently installed on the 3cx machine. Is this the problem?
 
Try to replace "localhost" to 192.168.1.7 in X-lite account configuration.
It is funny, but it can help in some cases :)
 
Hmmm... You caught me there. Didn't expect the 3CX machine to be the personal use machine also. Not that this is bad, unless you are a gamer or graphics artist :)

No, having the softphone on the 3CX server should not be the cause to the best of my knowledge. Would have been nice to have another extension to test local vs external calls though. When dialing voicemail you just key in 999? Did you make any entries in the X-Lite setup other than the basic SIP account first page?
 
replacing localhost with 192.168.1.7 did it. Thanks a lot. Strange that external calls worked though while it was set to localhost!
 
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