Connect to IPAustria

[Fohnbit]

Hello

I get several Infos to connect to IP Austria, but it doesn´t work.
I can receive calls, but not set an call out.
IPAustria route all numbers to me and I have rules for the extension.

I mean the CALL-ID is not correct:
This is from Asterisk and work:

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.200.13:5060;branch=z9hG4bK2d3abb18;rport
Max-Forwards: 70
From: "431235023410" <sip:[email protected]>;tag=as60783539
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.15.1
Date: Mon, 22 Apr 2013 09:11:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 1044607301 1044607301 IN IP4 192.168.200.13
s=Asterisk PBX 1.8.15.1
c=IN IP4 192.168.200.13
t=0 0
m=audio 18016 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

If I call, I have the Wireshark attached.

I see, that the Call_ID is different. At the Asterisk log, he append the domain "@node1.ipaustria.at"

Can someone help me to configure 3CX?

Thank you!

 

[lneblett]

Please post a 3CX log so that we can see the issue.

 

[Fohnbit]

05-Aug-2013 15:19:05.807   Leg L:3.1[Extn] is terminated: Cause: BYE from PBX
05-Aug-2013 15:19:05.807   [CM503008]: Call(C:3): Call is terminated
05-Aug-2013 15:19:05.799   [CM503023]: Call(C:3): Call recording is stopped, audio file: C:\ProgramData\3CX\Data\Recordings\122\[H*******]_122-004366*******58_20130805151900(3).wav
05-Aug-2013 15:19:05.797   Leg L:3.3[EndCall] is terminated: Cause: BYE from 127.0.0.1:40600
05-Aug-2013 15:19:00.267   [CM503007]: Call(C:3): EndCall:EndCall has joined, contact <sip:[email protected]:40600>
05-Aug-2013 15:19:00.264   [CM503007]: Call(C:3): Extn:122 has joined, contact <sip:[email protected]:5060>
05-Aug-2013 15:19:00.264   [CM503022]: Call(C:3): Call recording is started, audio file: C:\ProgramData\3CX\Data\Recordings\122\[Han******ing]_122-00436******58_20130805151900(3).wav
05-Aug-2013 15:19:00.258   L:3.3[EndCall] has joined to L:3.1[Extn]
05-Aug-2013 15:19:00.063   [CM503025]: Call(C:3): Calling T:EndCall:EndCall@[Dev:sip:[email protected]:40600;rinstance=35e78234f3f71406] for L:3.1[Extn]
05-Aug-2013 15:19:00.055   Leg L:3.2[Line:10002>>004366*****8] is terminated: Cause: 404 Not Found/INVITE from 213.164.25.150:5060
05-Aug-2013 15:19:00.055   L:3.1[Extn] failed to reach Line:10002>>00436******8, reason Not Found
05-Aug-2013 15:19:00.055   Call to T:Line:10002>>004366******8@[Dev:sip:18*****[email protected]:5060] from L:3.1[Extn] failed, cause: Cause: 404 Not Found/INVITE from 21******50:5060
05-Aug-2013 15:19:00.053   [CM503003]: Call(C:3): Call to <sip:00436******[email protected]:5060> has failed; Cause: 404 Not Found/INVITE from 21*******50:5060

I mark IP Addresses and telephone numbers with * to keep privacy

Thanks

 

[lneblett]

It is always helpful when trying to compare a working environment to a non-working one that the test parameters be the same and the resultant data logs cover the same points.

The logs from each show different things, leaving me to guess -
1. Did you run the firewall checker and did it pass? I assume that the 3CX and Asterisk packages are running on different machines and as a result have different IP addresses. I can see the Asterisk internal IP, but not the 3CX, so I question if port forwarding has been adjusted and tested; both use port 5060.

In the case of the Asterisk post you dialed - To: <sip:[email protected]>

In the case of 3CX you dialed: Call to <sip:00436******[email protected]:5060>

Am uncertain why you blocked the number from being seen on one log and then not the other. Because the numbers are different, I do not know if the 3CX dialed number is a valid format or not. On the one hand, I might question why a "00" lead in rather than a "0" as the Asterisk post shows; one is 11 digits, the other 12.

 

[Fohnbit]

Hello

The Asterisk Code is from the VOIP provider. It is not my enviroment.
Firewall is all fine!

I can made calls inbound.
With an direct Login (only one number) I can made also calls. But then I have for each extension an own login and it is also expensive.
So I want to change to an "open" login (don´t know the english word, when all outbound extension are receiving). With this, I have only one login and have to handle the DID.
With the "open" Login the inbound works, but not the outbound. Because with this account, the Provider has to check my call-number (extension) ... and this seems the problem.

The CALL-ID in the log of Wireshark are different.
At the Asterisk System it have the ending of the VOIP Provider ... is this important?

I dont´t want to post my telephone Number and also nocht the account number.
The destination call number is right. I checked it. In this example it was recording the international code (0043) for Austria. In the Asterisk example it is the national code (without 0043).
I tested both, always the same.

Have someone an Idea?

Thanks

 

[lneblett]

Sorry, but the log you posted is not enough. I can see the error "not found", but I do not see how the call was initiated nor how the call was routed via any outbound rule.

I can also see that it attempted to reach L10002, but without knowing the rules or how the provider is setup in VoIP Provider, I am guessing.

I see that one log simply shows a number and the other shows the number and the destination qualifier of node1.ipaustria.at. IS this the issue you refer to as the caller-ID not being the same? Presumably, under the VoIP provider tab, you have set-up the provider with a name or IP, port, an outbound proxy and port as well. Hopefully the provider supports registrations and you have registration set to required for both in and outbound calls and the registration indicator is green, If not set as indicated, the indicator will be green as you have instructed the system to ignore and therefor a red indicator (meaning not registered) is of no value.
With regard to how the outbound parameters are setup, the provider needs to instruct as to what is needed (remote party ID, To: Host part, etc.). Did you use one of the templates like generic SIP provider. Is "GWhostport gateway/provider host/port" specified in any of your outbound parameters?