Source ID Problem which could not be solved

[macher]

HI,

we have installed the software and outbound works well.
But Inbounded just without the extensions.
I have checked the part from the http://www.3cx.com/support/source-id-errors.html
But this did not help because 018903642200 is the main number 018903642 plus extention 200
WOuld I need to do this for every extention?
This I thing wouldn´t make a sense.

Your help is appretiated - thanks


[CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected]:5060;rinstance=035e9f46a90a9e71 SIP/2.0
Via: SIP/2.0/UDP 212.41.253.181;branch=z9hG4bK1997.80752825.0
Via: SIP/2.0/UDP 172.29.183.11;rport=5060;branch=z9hG4bK1997.3c622557.0
Via: SIP/2.0/UDP 172.29.183.103:5062;branch=z9hG4bK7b5cd19aa
Max-Forwards: 69
Record-Route: [sip:212.41.253.181;r2=on;lr=on;ftag=deb5771a5389633]
Record-Route: [sip:172.29.183.10;r2=on;lr=on;ftag=deb5771a5389633]
Record-Route: [sip:172.29.183.11;lr=on;ftag=deb5771a5389633]
Contact: [sip:[email protected]:5062]
To: [sip:[email protected]]
From: [sip:[email protected]];tag=deb5771a5389633
Call-ID: [email protected]
CSeq: 441193280 INVITE
Session-Expires: 600
Supported: timer, replaces
User-Agent: Patton SN2400 MxSF v3.2.8.45 00A0BA020B1D R3.21 2007-09-14_RFE10808 H323 SIP
P-Preferred-Identity: [sip:[email protected]]
Content-Length: 0

Thanks in Advance,
Ernst

 

[Powermage]

Add an DID number with that caller id and route it to an extension/que/group

 

[macher]

HI, I have tried this -

I have added the following DID:

DID Name: Test 200
DID/MSN/Mask: 018903642200
Override incoming Caller ID with DID name: Selected

Anywhere

Connect to Extension: 200

And I still get the following Error Message in the Server Status

[CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected]:5060;rinstance=6c1a1652e42bdc79 SIP/2.0
Via: SIP/2.0/UDP 212.41.253.181;branch=z9hG4bK279f.ad903e17.0
Via: SIP/2.0/UDP 172.29.183.7;rport=5060;branch=z9hG4bK279f.921fc1a2.0
Via: SIP/2.0/UDP 172.29.183.102:5062;branch=z9hG4bK48deadcb7
Max-Forwards: 69
Record-Route: [sip:212.41.253.181;r2=on;lr=on;ftag=156009dabfd12b0]
Record-Route: [sip:172.29.183.10;r2=on;lr=on;ftag=156009dabfd12b0]
Record-Route: [sip:172.29.183.7;lr=on;ftag=156009dabfd12b0]
Contact: [sip:[email protected]:5062]
To: [sip:[email protected]]
From: [sip:[email protected]];tag=156009dabfd12b0
Call-ID: [email protected]
CSeq: 1468900846 INVITE
Session-Expires: 600
Supported: timer, replaces
User-Agent: Patton SN2400 MxSF v3.2.8.45 00a0ba00ac60 R3.21 2007-09-14_RFE10808 H323 SIP
P-Preferred-Identity: [sip:[email protected]]
Content-Length: 0

Did I fill in the fields wrong?

Thanks,
Ernst

 

[Powermage]

If u use an * as the DID and select the voip provider, what do you get then?

 

[macher]

Where would I use the * ?

Could you show me how it should look like?

 

[macher]

I have tried it in the following way

DID Settings

Name 510
DID/DDI number/mask *510*
Override incoming Caller ID with DID name checked

VOIP Provider: Inode

Connect to Extension: 510

Still the same problem


11:29:23.609 evt::CheckIfAuthIsRequired::not_handled [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected]:5060;rinstance=8f12b17e4b3db109 SIP/2.0
Via: SIP/2.0/UDP 212.41.253.181;branch=z9hG4bKfb58.2ec0d162.0
Via: SIP/2.0/UDP 172.29.183.15;rport=5060;branch=z9hG4bKfb58.28ff9ab2.0
Via: SIP/2.0/UDP 172.29.183.103:5062;branch=z9hG4bK7bb16401b
Max-Forwards: 69
Record-Route: [sip:212.41.253.181;r2=on;lr=on;ftag=40f06e1165d00fc]
Record-Route: [sip:172.29.183.10;r2=on;lr=on;ftag=40f06e1165d00fc]
Record-Route: [sip:172.29.183.15;lr=on;ftag=40f06e1165d00fc]
Contact: [sip:[email protected]:5062]
To: [sip:[email protected]]
From: [sip:[email protected]];tag=40f06e1165d00fc
Call-ID: [email protected]
CSeq: 1716034216 INVITE
Session-Expires: 600
Supported: timer, replaces
User-Agent: Patton SN2400 MxSF v3.2.8.45 00A0BA020B1D R3.21 2007-09-14_RFE10808 H323 SIP
P-Preferred-Identity: [sip:[email protected]]
Content-Length: 0

11:29:23.609 evt::CheckIfAuthIsRequired::not_handled [CM302001]: Authorization system can not identify source of: SipReq: INVITE [email protected]:5060 tid=fb58.2ec0d162.0 cseq=INVITE [email protected]:5062 / 1716034216 from(wire)

 

[archie]

Open Source identification section for Patton gateway and set following:
SIP Field | Value | Custom Value
Request Line URI : Host part | Custom Field | 172.29.183.103:5060

 

[macher]

Hi,

I have asked the provider to do so - but they did tell me they can´t because all of their customers are on the same patton box and so it could be a problem for the others.

Any other Idea - I would like to stick with 3cx and not go back to asterisk - because I like the way the system looks like

But I need to make it working

best regards,
Ernst

 

[archie]

Sorry, do you have administrator's rights for you 3CX PBX box or not? I was speaking about settings of 3CX box. That Patton is not in your posession? Is it on some VoIP provider's site?

 

[macher]

Hi,

our configurations is as follow

we have a telecom provider named Inode and we have there the Telenode TITAN Package which works with SIP Protokol.

On our Side we have a PC with the 3CX Software which we did configure using the base settings for the Generic VOIP Provider.

The provider sends us the calls directly over VOIP with the SIP Protokol to our 3CX Telefon system.

We do not have any Patton Box inbetween.

And the connections works because we can call out.

But unfortunatly not in!

Thanks,
Ernst

 

[archie]

Than open "Edit VOIP Provider" page for your provider in 3CX Management console and in section Source identification apply the changes I've mentioned above

 

[macher]

Thanks - it works!!!

The only question now still is how can I adress the extensions.
Every call which comes in goes to the routed inbound extension.

Chears,
Ernst

 

[h2009]

Sorry can you clarify what you want the incoming call to do please. Thanks.

 

[macher]

Good morning!

Yes of course.

We have 7 extensions for our phones and one fax line.

This extension are Numbered from 200 to 206.
We would like that our customers can directly call trough to the needed person.
And if someone calls the headnumber it should ring at just one extension.

i.e. if someone calls 01 890 4236 or 01 890 4236 he should come to extension 200

or if someone calls 01 890 4236 205 or 01 890 4236 203 he should come to the called person.
or if he calls 01 890 4236 150 he should be connected to the telefax machine.

So people should be able to call trough.
The SIP line incorporates the requested extension

like
INVITE sip:[email protected]:5060;rinstance=035e9f46a90a9e71 SIP/2.0
or
INVITE sip:[email protected]:5060;rinstance=8f12b17e4b3db109 SIP/2.0

Thanks,
Ernst

 

[FITEC]

i.e. if someone calls 01 890 4236 or 01 890 4236 he should come to extension 200

noticed that 0 is usually appended when someone calls the headnumber (without extension) - try to set SIP-ID 0 at the receptionist's extension

or if someone calls 01 890 4236 205 or 01 890 4236 203 he should come to the called person.
or if he calls 01 890 4236 150 he should be connected to the telefax machine.

check if the SIP-ID at the extension is set to 205, 203 or 150

 

[mickp]

i.e. if someone calls 01 890 4236 or 01 890 4236 he should come to extension 200

noticed that 0 is usually appended when someone calls the headnumber (without extension) - try to set SIP-ID 0 at the receptionist's extension

or if someone calls 01 890 4236 205 or 01 890 4236 203 he should come to the called person.
or if he calls 01 890 4236 150 he should be connected to the telefax machine.

check if the SIP-ID at the extension is set to 205, 203 or 150

Um. I don't think sip id is what you want to be looking at. Sip id is for internet calls with [email protected] style calls afaik.

Mick.

 

[mickp]

Good morning!

Yes of course.

We have 7 extensions for our phones and one fax line.

This extension are Numbered from 200 to 206.
We would like that our customers can directly call trough to the needed person.
And if someone calls the headnumber it should ring at just one extension.

i.e. if someone calls 01 890 4236 or 01 890 4236 he should come to extension 200

or if someone calls 01 890 4236 205 or 01 890 4236 203 he should come to the called person.
or if he calls 01 890 4236 150 he should be connected to the telefax machine.

So people should be able to call trough.
The SIP line incorporates the requested extension

like
INVITE sip:[email protected]:5060;rinstance=035e9f46a90a9e71 SIP/2.0
or
INVITE sip:[email protected]:5060;rinstance=8f12b17e4b3db109 SIP/2.0

Thanks,
Ernst

If I understand your request correctly, you have a working sip trunk and just need to configure some did (direct in-dial) numbers. My appologies if I'm missing the point and there's some other issue. Review the instructions http://www.3cx.com/manual/3CXPhoneSystemManual6/phone-system47.html and see how you go.

Mick.

 

[FITEC]

Um. I don't think sip id is what you want to be looking at. Sip id is for internet calls with [email protected] style calls afaik.

you're absolutely right, Mick, The SIP-ID is used for internet calls but in this scenario it seems that the sip-provider uses a different field for the DIDs. will check if it's related to Source identification section (http://www.3cx.com/forums/source-id-problem-which-could-not-be-solved-4819.html#p38255) and let you know.

If I understand your request correctly, you have a working sip trunk and just need to configure some did (direct in-dial) numbers. My appologies if I'm missing the point and there's some other issue. Review the instructions http://www.3cx.com/manual/3CXPhoneSystemManual6/phone-system47.html and see how you go.

DID does not work in this scenario - but in any case it's related to the SIP-provider because it works with a different one :x