3CX V11 issue when divert call from Digital receiption

[zhangjx99]

Hi there,
I just switched over to V11 and import previous configuration, everything ok but faced issue when divert from digital receipts, it can't find the outbound rules. eg. message saying: Destination can't be found, nor routing rule configured.

in fact I have outbound rules with it.

JX

 

[lneblett]

Can you provide more info? Need to understand how the digital receptionist is configured, how the outbound rule for the given keys and/or extensions are configured, office hours or other rules that may be in effect, and if your provider and/or gateway are registered. I assume this is for external calls coming in and that internal calls going out are ok (implying that your provider/gateway are working).

 

[zhangjx99]

My setup is as below:

1. Grandstream 8 ports FXO gateway to link a traditional PBX via 4 FXS extensions (disabled another 4 ports);
2. Outbound rule:
Prefix: 28
Route 1: FXO gateway, strip digits: 2
Route 2: (it is the same even delete it)
3. When I initial the call from
a) internal IP Phone, for example: 28133, it will route to destination perfectly;
b) if call from outside, or from internal call 80000(digital receipts), and followed by 28133, call failed;
c) further testing, when I setup 28000 to forward all call to 28133, and re-test b) steps, by key in 28000, it will route to destination correctly.

there is no such issue with 3CX V8 (my live system).
I tried to install V10, but is the same issue. (I can't use V9 due to FXO gateway registration problem).
License: 16SC

Your help will be much appreciated.
Thanks.
JX

 

[leejor]

So, the numbers, beginning with 28, work, if dialled from an extension?

Can you post the actual 3CX log, of a failed call,(verbose mode if possible).

 

[SY]

Open Settings->Advanced and then Custom Parameters tab.
Find (using Filter field) IVRNOCHECKFORDN parameter.
set its value to 1.
Press Apply or OK at the bottom of page.
Restart is not required.
After this change, IVR will allow calls to any 5 digits number even if it does no exist in the system.

 

[zhangjx99]

Hi Stepan,

Yes, it works perfectly.
I guess it is the new feature after V8 as I can't find it from V8.

Thank you very much for the advice.

JX

 

[zhangjx99]

Hi Stepan and all,

I would like to re-open this topic to seek for helping.

my setup is like that:
Site A: 22161 is from NEC IP phone system,
IP: 10.101.4.238,
SIP Trunk to 3CX (site B), Auth. ID: 27100
Outbound Rules to divert prefix 6xxxx to Site B
Outbound Rules to divert prefix 21xxx to Site B
Site B: VOIP system: 3CX system
IP: 10.101.4.6
3CX extension: 21161
SIP Trunk to (Site A), Auth. ID: 27100
SIP Trunk to (Site C), Auth. ID: 21008
Outbound Rules to divert prefix 65xxx to Site C
Outbound Rules to divert prefix 22xxx to Site A
Site C: 65299 is from Asterisk IP phone system,
IP: 192.108.8.97,
SIP Trunk to 3CX (Site B), Auth. ID: 65008
Outbound Rules to divert prefix 2xxxx to Site B

Now, when I call from 65299(Site C), no any issue to call 21161(Site B) or 22161(Site A), however, when call from 22161(Site A), I am able to reach 21161 (extension in Site B), but fail to reach 65299(Site C).

3CX log activities:
05-Jul-2013 11:23:35.195 [CM102001]: Authentication failed for AuthFail Recv Req INVITE from 10.101.4.238:5060 tid=-317351d6acff0-245 [email protected]:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.101.4.238:5060;branch=z9hG4bK-317351d6acff0-245
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: "22161"<sip:[email protected]:5060>;tag=661751d6acfe5-245
Call-ID: [email protected]
CSeq: 2 INVITE
Session-Expires: 1800
Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK
Content-Type: application/sdp
Proxy-Authorization: Digest username="27100",realm="3CXPhoneSystem",nonce="414d535c07e6cd3792:13566155bd07d2578b6f8329dd68bbb1",uri="sip:[email protected]:5060",response="8cfd52b80b5af79dbe818f60df82416e",opaque=""
Supported: timer
Content-Length: 175

v=0
o=IPS 20855 0 IN IP4 10.100.4.243
s=IPS
c=IN IP4 10.100.4.243
t=0 0
m=audio 10118 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
; Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings
IVRNOCHECKFORDN = 1 already.

Please help, what is the Credentials mentioned in above log file?

thanks.
JX