[startelecom]

Provider responds with ACK to 192.168.2.60:5095 which never makes it to the 3CX. Why isn't 3CX using the public (STUN detected) IP in the Contact field?

Note: This is on inbound calls. So 8551234567 is my DID.

X.Y.Z.A = Source IP
X.Y.Z.b = Provider IP



U 2011/03/12 15:05:20.459041 X.Y.Z.A:5095 -> X.Y.Z.b:5094
SIP/2.0 200 OK.
Via: SIP/2.0/UDP X.Y.Z.b:5094;rport=5094;branch=z9hG4bKH254D4rH78j3a.
Contact: <sip:8551234567@192.168.2.60:5095>.
To: <sip:8551234567@ X.Y.Z.15:5095>;tag=af65267a.
From: "I ROBOT"<sip:9051234567@X.Y.Z.b>;tag=Z20SBa3mBmt5r.
Call-ID: d4ed42e9-c786-122e-848c-00259010ee66.
CSeq: 9644081 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE.
Content-Type: application/sdp.
Supported: replaces.
User-Agent: 3CXPhoneSystem 9.0.15776.0.
Content-Length: 271.
.
v=0.
o=3cxPS 128731578368 464930209793 IN IP4 192.168.2.60.
s=3cxPS Audio call.
c=IN IP4 192.168.2.60.
t=0 0.
m=audio 7000 RTP/AVP 0 8 101.
c=IN IP4 192.168.2.60.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=sendrecv.

[abc123]

On the Edit VOIP provider > Advanced tab there is a settting to say which ip to provide in the contact information.

Your choices should be STUN, Internal or Specified. I think you accidentally (or something accidentally) clicked internal.

I strongly suggest you put in a static public ip in there. use STUN as a last resort.

[startelecom]

Hi Mark,

Thanks for your response. I checked the settings and "External(STUN resolved)" is currently selected. I will try using "Specified IP" option to see if it makes a difference.

I should add... this is an IP Authenticated trunk, no registrations for either inbound or outbound.

[startelecom]

Hi Mark,

Using "Specified IP" did not make a difference. 3CX is still sending the internal IP in the Contact field.

[abc123]

That was my fault.

I thought the 200 was response to register but it is a response to the invite request. The setting i gave you was for registration but as you said you are using ip based then it is unnecessary (though i would keep it as the external ip).

We have a couple of options to try here.

on the voip provider (broadvox?) Oubound parameters, what is the Contact:Host Part currently set to?

[SY]

Hi startelecom,

PBX log may explain some aspects of the problem. Could you please post a quotation from "Server activity Log" page (management console) which will show the flow of a problematic call?

Thanks

[startelecom]

Sure thing!

One thing to note - the provider has multiple addresses from which the call could originate, in this case it comes through X.Y.Z.B, whereas only one IP is set up on the trunk for outbound purposes - X.Y.Z.C.


10:18:30.546 [CM503008]: Call(11): Call is terminated
10:18:30.546 [CM503021]: Call(11): ACK is not received
10:18:11.375 [CM503007]: Call(11): Device joined: sip:205@192.168.2.106:5060;transport=udp
10:18:11.375 [MS210002] C:11.1:Offer provided. Connection(transcoding mode): 192.168.2.60:7056(7057)
10:18:11.375 [MS210001] C:11.3:Answer received. RTP connection[unsecure]: 192.168.2.106:3000(3001)
10:18:11.375 Remote SDP is set for legC:11.3
10:18:10.906 [CM505001]: Ext.205: Device info: Device Identified: [Man: Aastra;Mod: 9133i;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, recvonly] UserAgent: [Aastra 9133i/1.4.2.3000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45] PBX contact: [sip:205@192.168.2.60:5095]
10:18:10.906 [CM503002]: Call(11): Alerting sip:205@192.168.2.106:5060;transport=udp
10:18:10.656 [CM503025]: Call(11): Calling Ext:Ext.205@[Dev:sip:205@192.168.2.106:5060;transport=udp]
10:18:10.656 [MS210002] C:11.3:Offer provided. Connection(transcoding mode): 192.168.2.60:7060(7061)
10:18:10.593 [CM503004]: Call(11): Route 1: Ext:Ext.205@[Dev:sip:205@192.168.2.106:5060;transport=udp]
10:18:10.593 [CM503010]: Making route(s) to <sip:205@127.0.0.1:5095>
10:18:10.593 [CM505003]: Provider:[Generic SIP Trunk -] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [FreeSWITCH-mod_sofia/1.0.6-svn-exported] PBX contact: [sip:pbx@192.168.2.60:5095]
10:18:10.593 Refer: from=<sip:800@127.0.0.1:5095>;tag=c1410750; to="8551234567:6471234567"<sip:6471234567@127.0.0.1:5095>;tag=60737f71; RefTo=<sip:205@127.0.0.1:5095>
10:18:10.375 [MS210003] C:11.2:Answer provided. Connection(transcoding mode[unsecure]):127.0.0.1:7058(7059)
10:18:10.375 [MS210000] C:11.2:Offer received. RTP connection: 127.0.0.1:40626(40627)
10:18:10.375 Remote SDP is set for legC:11.2
10:18:06.390 [MS211000] C:11.1: X.Y.Z.B:21306 is delivering DTMF using RTP payload (RFC2833). In-Band DTMF tone detection is disabled for this call segment.
10:18:05.203 Currently active calls - 1: [11]
10:17:58.468 [CM503007]: Call(11): Device joined: sip:800@127.0.0.1:40600;rinstance=210059fd056f8681
10:17:58.468 [CM503007]: Call(11): Device joined: sip:pbx@X.Y.Z.C:5094
10:17:58.468 [MS210003] C:11.1:Answer provided. Connection(transcoding mode[unsecure]):192.168.2.60:7056(7057)
10:17:58.453 [MS210001] C:11.2:Answer received. RTP connection[unsecure]: 127.0.0.1:40626(40627)
10:17:58.453 Remote SDP is set for legC:11.2
10:17:58.453 [CM505001]: Ext.800: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX IVR;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX IVR] PBX contact: [sip:800@127.0.0.1:5095]
10:17:58.453 [CM503002]: Call(11): Alerting sip:800@127.0.0.1:40600;rinstance=210059fd056f8681
10:17:58.312 [CM503025]: Call(11): Calling Ext:Ext.800@[Dev:sip:800@127.0.0.1:40600;rinstance=210059fd056f8681]
10:17:58.312 [MS210002] C:11.2:Offer provided. Connection(transcoding mode): 127.0.0.1:7058(7059)
10:17:58.250 [MS210000] C:11.1:Offer received. RTP connection: X.Y.Z.B:21306(21307)
10:17:58.250 Remote SDP is set for legC:11.1
10:17:58.250 [CM503004]: Call(11): Route 1: Ext:Ext.800@[Dev:sip:800@127.0.0.1:40600;rinstance=210059fd056f8681]
10:17:58.250 [CM503010]: Making route(s) to <sip:800@192.168.2.60:5095>
10:17:58.250 [CM505003]: Provider:[Generic SIP Trunk -] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [FreeSWITCH-mod_sofia/1.0.6-svn-exported] PBX contact: [sip:pbx@192.168.2.60:5095]
10:17:58.250 [MS210000] C:11.1:Offer received. RTP connection: X.Y.Z.B:21306(21307)
10:17:58.250 Remote SDP is set for legC:11.1
10:17:58.250 [CM503001]: Call(11): Incoming call from 6471234567@(Ln.10000@Generic SIP Trunk -) to <sip:800@192.168.2.60:5095>
10:17:58.234 [CM503012]: Inbound any hours rule (8551234567) for 10000 forwards to DN:800
10:17:58.234 Looking for inbound target: called=8551234567; caller=6471234567
10:17:58.234 [CM500002]: Info on incoming INVITE:
INVITE sip:8551234567@X.Y.Z.A:5095 SIP/2.0
Via: SIP/2.0/UDP X.Y.Z.B:5094;rport=5094;branch=z9hG4bK5Kc8Sp7t5QBeD
Max-Forwards: 67
Contact: <sip:mod_sofia@X.Y.Z.B:5094>
To: <sip:8551234567@X.Y.Z.A:5095>
From: "6471234567"<sip:6471234567@X.Y.Z.B>;tag=UHyHy8U9DXS1r
Call-ID: 89724be5-c81f-122e-13b3-00259010ee68
CSeq: 9676875 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
Allow-Events: talk, refer
Privacy: none
P-Asserted-Identity: "6471234567" <sip:6471234567@X.Y.Z.B>
Content-Length: 0
P-hint: outbound->inbound
X-PeerUuid: 3000
X-SrcIp: X.Y.Z.B
X-FS-Support: update_display

[SY]

The strange thing: PBX thinks that the address X.Y.Z.B is resided in the local network.
Do you have any additional hints? (network topology, network layer configuration of the host, PBX configuration etc.)

Thanks

[startelecom]

Nothing really strange.

The server itself - one NIC with the internal IP listed in the capture 192.168.2.60

The server is behind a linksys router which itself has one external IP X.Y.Z.A. No SPI or anything fancy as far as I know.

Couple of switches in between... again nothing too fancy.

[startelecom]

anyone?

[SY]

anyone?

straight procedure to get straight answer:
1. Make a wireshark capture of the network traffic (on the 3CX PhoneSystem host) for the problematic call.
2. post it here.

Thanks

[smb1]

On the linksys router, do you have SIP ALG turned on?

I always disable this on any router being used to process SIP traffic.

[startelecom]

I see the bug still exists in v10. It's simply impossible to force 3CX to use the public IP.

I have tried:

1. Changing Providers>Advanced>Registration Settings > Specified IP (using my public IP)
2. Settings > Netowrk > STUN Server > Turn Off Stun Server > Public IP to specify in Contact and SDP (using my public IP)


Note regarding #1: I am using IP authentication with the provider server, so my 3CX does not actually register. So I am not sure if the setting effectively changes anything.

[netswork]

Startelecom:

To reiterate what David said above, you need to look at your router for the cause. Newer routers, the ones that do more than just route, have built in technology to "help" your sip traffic. If you do a wireshark from inside the network, and then from the outside of the router you will likely see were the problem is. Your router is over-writing the field.

I have seen the same thing many times with fortigate firewalls. You need to make sure and disable all sip helper features. In fortigates this can only be disabled via the command line. Which model of linksys are you using?


You could do a quick test by removing your router and plugging your phone system directly to the internet to make one quick test call.

[startelecom]

I am using Linksys WRT54GL, running Tomato Version 1.28. I don't think it does any packet inspection or ALG.

More accurately, some versions of Tomato have SIP helper, but the version I am using does not. Just to be safe, I disabled the H323 helper, and the results are still the same. The contact address is the internal IP.