Vodafone Greece Sip trunk issues

[MichalisE]

Hi,
I am trying to set up for my company 3CX with Vodafone Greece. I have a lot of problems with them to provide me the needed information for the sip trunk.

After 3 days of calls and endless discussions I managed to set up the trunk with authentication type as “Do not require - IP Based”, but I have the username/pass Vodafone gave me on the fields.

The inbound calls are working fine, but on the outbound calls when the other side answers the call is dropping.

Error message on 3CX log says that call or registration failed and replied 403 Forbidden (bad auth; from IP ***)

Any idea of what might causing this? If anyone worked with Vodafone GR before please help me.

We are using 3CX v16.0.1.273 on Windows Server 2012 r2.
P.S I am not even sure that I have the correct Gateway. But I suppose it wouldn’t work at all if was incorrect.

 

[YiannisH_3CX]

Hello @MichalisE

It sounds like the trunk should be authentication based as you have a username and a password and from the error you are getting during outbound calls.
Unfortunately Vodafone in Greece is not a supported provider so we cannot know for sure how to configure them on the PBX as they were never tested.

What happens when you try to register the trunk? Do you get an error from Vodafone?

There a similar thread that might assist you set the trunk up:
https://www.3cx.com/community/threads/vodafone-greece-sip-trunk.60378/

 

[MichalisE]

Thanks for your reply Yianni.
I saw that thread and it didn't help. There is no authentication, I spoke with them and they told me that they disable authentication. That's why I have it on "Do not require - IP Based".

Do you know or someone else the information I need from them? So we cross-check what I have.
I know that a lot managed to set up 3CX with Vodafone GR and I spoke with two of your partners and they asked around €100/hr to set it up for me. That's outrageous.

 

[eddv123]

Can you run a PCAP trace from 3CX:
https://www.3cx.com/docs/capture-network-traffic/

And make a test call.

This will give you a definitive answer as to whether the provider is actually responding or not - you are getting a 403 Forbidden which is normally a client side error telling me (at least on the surface) that the settings on the PBX are incorrect.

 

[YiannisH_3CX]

You can wait and see if another forum user has setup Vodafone and is willing to help or you can hire someone that can do this for you. How much you are willing to pay someone depends on how much you value your time. If you spend another 5 hours on this and a 3CX partner needs 20 minutes to solve your issue then it might be worth it. That is up to you.
For now i would recommend running a pcap like @eddv123 suggested. Leave the trunk to IP based and see if Vodafone is challenging your outbound Invite with a 407 authentication required message.

 

[MichalisE]

Ok guys @YiannisH_3CX @eddv123 !! Thanks for your response!

 

[MichalisE]

Do you know if Port 443 has anything to do with the outbound calls?
I just notice that is closed.

 

[YiannisH_3CX]

That is your HTTPS port (if you used that during the installation) and will affect SIP Calls so it has nothing to do with calls. You need it for different purposes though like accessing the management console from outside or provisioning remote devices or clients.

 

[MichalisE]

Ok, thanks!

 

[MichalisE]

Hi again,
I run Wireshark check and I got this message.
Any idea of what might goes wrong?

Session Initiation Protocol (BYE)
Request-Line: BYE sip:216000*@SIP CLIENT IP:5060 SIP/2.0
Method: BYE
Request-URI: sip:216000*@SIP CLIENT IP:5060
Request-URI User Part: 216000*
Request-URI Host Part: SIP CLIENT IP
Request-URI Host Port: 5060
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP SIP HOST IP:5060;branch=z9hG4bK418f052f;rport
Transport: UDP
Sent-by Address: SIP HOST IP
Sent-by port: 5060
Branch: z9hG4bK418f052f
RPort: rport
From: <sip:003579910*@SIP HOST IP>;tag=as0826e6de
SIP from address: sip:003579910*@SIP HOST IP
SIP from address User Part: 003579910*
SIP from address Host Part: SIP HOST IP
SIP from tag: as0826e6de
To: "216000"<sip:216000@SIP HOST IP>;tag=2d74b670
SIP Display info: "216000*"
SIP to address: sip:216000*@SIP HOST IP
SIP to address User Part: 216000*
SIP to address Host Part: SIP HOST IP
SIP to tag: 2d74b670
Call-ID: yesFZupcl03y51g-oWdyBw..
[Generated Call-ID: yesFZupcl03y51g-oWdyBw..]
CSeq: 102 BYE
Sequence Number: 102
Method: BYE
User-Agent: fw2018100112-IVV34200.N3A2UHM
Max-Forwards: 70
X-Asterisk-HangupCause: Unknown
[Expert Info (Note/Undecoded): Unrecognised SIP header (x-asterisk-hangupcause)]
[Unrecognised SIP header (x-asterisk-hangupcause)]
[Severity level: Note]
[Group: Undecoded]
X-Asterisk-HangupCauseCode: 10
[Expert Info (Note/Undecoded): Unrecognised SIP header (x-asterisk-hangupcausecode)]
[Unrecognised SIP header (x-asterisk-hangupcausecode)]
[Severity level: Note]
[Group: Undecoded]
Content-Length: 0

 

[eddv123]

What are you connecting this to, another PBX or something ?
X-Asterisk-HangupCause: Unknown

http://forums.asterisk.org/viewtopic.php?f=1&t=72428

Could be IAX trunking, however it has been a few years since I was involved with Asterisk.

In Wireshark go to Telephony >> VoIP calls >> SIP/Call flow.

What you are looking for is to see where the response code is coming from. If you are not familiar with Wireshark PM me the PCAP trace and I will look at it for you - let me know the parties involved as well.

 

[MichalisE]

Hey @eddv123, thanks for your response.
I’ve checked what you said and I can see the call. On the comments column it says "INVITE 407 200”.

Any idea of what that means?

 

[eddv123]

Without seeing the trace that sounds like the INVITE message being sent from the PBX to provider - what you want is the message sent back (you can see on the SIP flow arrows denoting responding messages).

 

[MichalisE]

@eddv123 Can I have your email to send you the pcap trace?

 

[eddv123]

PM it to me on here and I will take a look.

 

[eddv123]

Looking at your PCAP there is no 403 forbidden and it looks like the call is taking place now (so the original error seems to be sorted I think) please advise however of the current behaviour.

What is odd is the order of messages you are getting.

* You have an INVITE, 407 unauthorised, ACK, all the normal stuff you should see however x2 in relation to the message.

* The RTP stream initiates and then in the middle of it you have your 180 ringing, 200 OK and another RTP stream (but not 2 the second time around).

* the call example is between 10.10.0.3 and 10.10.0.2 - these look to be local class A addresses one is 3CX the other (which the user agent I dont recognise). What are you connecting to ? looks like a local SBC/layer 3 gateway device.

The only time I have seen SIP messages displaying in this manner was when the trunk was traversing a firewall which had the wrong NAT configured and it was messing with the packets.

 

[MichalisE]

10.10.0.2 is the SIP Trunk IP and 10.10.0.3 is my 3CX (Server SIP network card),

FYI, my installation is like this: Vodafone router internet port -> EdgeRouter X -> Network switch -> Server (3CX), PCs, IP Phones
For SIP Trunk is like this: Vodafone router SIP port -> Server (SIP network card)

So what do you think is the problem? Something with Vodafone's firewall or SIP Trunk IP?

By the way, I can make an outbound call, but when answered the call drops.

 

[eddv123]

OK so you sound like you are double -NAT'ng here - correct ? are these 2 firewall/routers or is one of them a modem passing through to the router ? you need to see if you have 2 sets of NATing in place - this would explain the malformation of the SIP dialogue packets.

 

[MichalisE]

The difference is that I am passing the Internet through a router (EdgeRouter) and then to a network switch that I have everything connected on (including the Server - 3CX).
As for the SIP trunk, I am connecting it straight from my provider's modem (Vodafone GR) on the Server dedicated SIP network card I have.

 

[eddv123]

Well the one thing I can definitely confirm is the packets are coming in x2 and in some cases the wrong order, you are getting 2 way audio, SIP packets and then only 1-way audio, this suggests router/firewall issues however the trace sent only captures local traffic. It would be good to see what is happening on the WAN side - unless you are connecting locally via MPLS or directly to the provider.