Digital Receptionist

[aeigb]

Hello All,

i would like to create a digital receptionist extension, which has following configuration:
1 -> transfer call to an external number
2 -> transfer to voicemail for user 33
0 -> repeat prompt

My SIP-ISDN-Gateway (Lancom 1724) has the IP-Address 192.168.0.222 and works well. Outbound an Inbound-Calls work without problems. Also all internal calls.

For User 33, i'm create a voicemail and set, that all call will be forward to Outside Number 0123 (the first zero is for an external call).

When i call the extension 33 from another sip-client, my call will be transfered to the outside-number, but when i use the digital receptionist, the 3cx told me "call transfer failed".

Here is the log:
18:47:35.437 Call::RouteFailed [CM503014]: Call(27): Attempt to reach [sip:[email protected]] failed. Reason: Not Found
18:47:35.437 CallLeg:nFailure [CM503003]: Call(27): Call to sip:[email protected] has failed; Cause: 404 Not Found; from IP:192.168.0.222
18:47:35.328 CallLeg:nFailure [CM503003]: Call(27): Call to sip:[email protected] has failed; Cause: 404 Not Found; from IP:192.168.0.222
18:47:35.203 CallLeg:nFailure [CM503003]: Call(27): Call to sip:[email protected] has failed; Cause: 404 Not Found; from IP:192.168.0.222

Does anybode know, what's wrong with these settings?

Thanks in advanced

Alexander

 

[archie]

You told your PSTN gateway to dial number 123 on land-line. I suppose your gateway can not believe that you're living in country with 3-digits local (land-line) numbers.

 

[aeigb]

The correct Number is longer than 123

The Number has six digits, and my PSTN-Gateway told me, that there is an unknown user who wants to make a call.

When i call the number from my account (SIP 60), there's no problem, also when i call the ext 33 (there is a all-call-forward) to this number -> both calls will bell on the right phone.

 

[archie]

'123' has 3 digits. If you want to keep real numbers in secret and replacing it with something else - please, at least keep its size.
'404' usually means problem with destination, for unknown source (caller) there's 403 reply.

I have a strange feeling that I'm completely lost and do not understand what did you do.

 

[aeigb]

What i would like to do is ...

... to create a digital receptionist with three options:
0 -> repeat prompt
1 -> transfer call to an outside number 123456
9 -> transfer call to voicemail of ext 33

For Option 1 i tried to configure an extension, which forward all calls to an outside number. When i call this extension from another extension it works very well. But when i use the "call extension" option from the digital receptionist, the call doesn't work.

I hope, i could give you all the information you need for help me.

Alexander

 

[archie]

I've asked our test dept. to reproduce this issue. If they could - I will fix it soon.

 

[archie]

No, we can not reproduce that issue. All is working fine in our test environment.

 

[nb]

problems with 2 digit PBX and DR -> transfer to external

Hi guys

We have not managed to replicate this.
We have tried this with a 2 digit extension PBX. Transfer to external number works fine.

We have also performed alternate tests with a normal 3 digit extension PBX and bulk tested transfers to an external number - VOIP Provider and also PSTN calls - these worked flawlessly.

I cannot understand what 123 is. The logs are showing that 123 doesnt exist.

When I tried to forward to an external number that is incorrect I get these logs -

11:28:41.640 Call::Terminate [CM503008]: Call(243): Call is terminated

11:28:41.609 Call::RouteFailed [CM503014]: Call(243): Attempt to reach [sip:[email protected];user=phone] failed. Reason: Not Found

11:28:41.609 CallLeg:nFailure [CM503003]: Call(243): Call to sip:[email protected]:5060 has failed; Cause: 404 Not Found; from IP:66.23.129.253

11:28:40.236 MediaServerReporting::SetRemoteParty [MS210002] C:243.2:Offer provided. Connection(transcoding mode): 82.102.77.191:9016(9017)

Here I am trying to call to an outside number and the settings I made in the extension are 0 (for outbound rule) and a fictitious number that I know doesnt exist. These are the exact logs you are getting. Why dont you try your mobile number?

regards

Nik

 

[aeigb]

I tried several numbers

Hello,

i tried several numbers for the outside call, but no number works. The "123" is an example for the external number, but i tried my correct pstn destination and my mobile.

10:49:32.578 Call::RouteFailed [CM503014]: Call(19): Attempt to reach [sip:[email protected]] failed. Reason: Not Found
10:49:32.578 CallLeg:nFailure [CM503003]: Call(19): Call to sip:[email protected] has failed; Cause: 404 Not Found; from IP:192.168.0.222

When i call the extension from another extension, then the forward works well, but when i call the extension from the digital receptionist, the call failed.

I don't understand the reason, because for me it's a call to the extension 33, equal i use the digital receptionist.

Here is a link to two screenshots of my 3cx server for the digital receptionist and the extension.
http://www.alexander-eich.de/index.php?id=153

As outside number i tried all combination with/without 0 (for outside calls), and i tried to append @voip-domain, but nothing work.

 

[rpn84]

Hello,

I am having almost the same issue reported here. I have an extention that is not registered that is set to forward all calls to an outside number via my voip provider voipcheap. When I call the extention directly the call will transfer just fine. It is only when using the digital receptionist that I get a message "call transfer failed". To me this indicates an issue with how the digital receptionist transferrs the call.

This is what I get in the event log:

12:23:46.880 Call::Terminate [CM503008]: Call(2): Call is terminated

12:23:37.677 Call::RouteFailed [CM503014]: Call(2): Attempt to reach [sip:[email protected]] failed. Reason: Reason Unknown

12:23:37.677 CallLeg:nFailure [CM503003]: Call(2): Call to sip:[email protected]:5060 has failed; Cause: 400 Bad request; from IP:194.221.62.198:5060

12:23:36.959 CallCtrl:nSelectRouteReq [CM503004]: Call(2): Calling: VoIPline:10000@[Dev:sip:[email protected]:5060]

12:23:32.396 MediaServerReporting:TMFhandler [MS211000] C:2.1: 192.168.16.26:16400 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.

12:23:20.022 CallCtrl:nLegConnected [CM503007]: Call(2): Device joined: sip:[email protected]:5060

12:23:19.084 CallCtrl:nLegConnected [CM503007]: Call(2): Device joined: sip:

12:23:19.069 CallCtrl:nSelectRouteReq [CM503004]: Call(2): Calling: IVR:300@[Dev]

12:23:19.053 CallCtrl:nIncomingCall [CM503001]: Call(2): Incoming call from Ext.200 to [sip:[email protected]]

I have tried chaning the external number to dial out over my pstn line and this actually works fine. It is only when going through my voip provider that the call fails... Please help!

 

[Ricambi America]

1) Create the extension for your cellular number:
* I used "301" in my case
* Forward all calls is enabled. Forward to "91336xxxxxxx" where "9" is an arbitrary prefix that I just chose for no real reason. The 1336xxxxxxx is my actual cellular number.



2) Create a new rule, in my case called "Forward to Mobile" which looks for prefix 9 and STRIPS one digit. See pictures below. Basically, whenever a number with prefix "9" is dialed through 3CX, it uses that rule and strips the "9", thus actually calling 1336xxxxxxx my cell.





Believe me, I struggled with this too. It works.