Fax server not working

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djboxny

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I installed the Fax server, and set the corresponding DID number that i got from Skype SIP, That number goes dirrectly to the fax, The fax server apparently picks up when someone call the fax line number but nothing is heard at all. and according to the fax machines i tried it with they say that No one answered.

here is the Berbose Logs

02:26:02.781 [CM503008]: Call(1): Call is terminated
02:26:01.375 [MS210007] C:1.2:Answer provided. Connection(by pass mode): 63.209.144.201:26918(26919)
02:26:01.375 [MS210001] C:1.1:Answer received. RTP connection[unsecure]: 63.209.144.201:26918(26919)
02:26:01.375 Remote SDP is set for legC:1.1
02:26:01.218 Remote SDP is set for legC:1.2
02:25:59.625 [MS210007] C:1.2:Answer provided. Connection(by pass mode): 63.209.144.201:26918(26919)
02:25:59.625 [MS210001] C:1.1:Answer received. RTP connection[unsecure]: 63.209.144.201:26918(26919)
02:25:59.625 Remote SDP is set for legC:1.1
02:25:59.468 Remote SDP is set for legC:1.2
02:25:57.875 [MS210007] C:1.2:Answer provided. Connection(by pass mode): 63.209.144.201:26918(26919)
02:25:57.875 [MS210001] C:1.1:Answer received. RTP connection[unsecure]: 63.209.144.201:26918(26919)
02:25:57.875 Remote SDP is set for legC:1.1
02:25:57.718 Remote SDP is set for legC:1.2
02:25:56.125 [MS210007] C:1.2:Answer provided. Connection(by pass mode): 63.209.144.201:26918(26919)
02:25:56.125 [MS210001] C:1.1:Answer received. RTP connection[unsecure]: 63.209.144.201:26918(26919)
02:25:56.125 Remote SDP is set for legC:1.1
02:25:55.968 Remote SDP is set for legC:1.2
02:25:54.375 [MS210007] C:1.2:Answer provided. Connection(by pass mode): 63.209.144.201:26918(26919)
02:25:54.375 [MS210001] C:1.1:Answer received. RTP connection[unsecure]: 63.209.144.201:26918(26919)
02:25:54.375 Remote SDP is set for legC:1.1
02:25:54.218 Remote SDP is set for legC:1.2
02:25:52.625 [MS210007] C:1.2:Answer provided. Connection(by pass mode): 63.209.144.201:26918(26919)
02:25:52.625 [MS210001] C:1.1:Answer received. RTP connection[unsecure]: 63.209.144.201:26918(26919)
02:25:52.609 Remote SDP is set for legC:1.1
02:25:52.468 Remote SDP is set for legC:1.2
02:25:50.875 [MS210007] C:1.2:Answer provided. Connection(by pass mode): 63.209.144.201:26918(26919)
02:25:50.875 [MS210001] C:1.1:Answer received. RTP connection[unsecure]: 63.209.144.201:26918(26919)
02:25:50.875 Remote SDP is set for legC:1.1
02:25:50.718 Remote SDP is set for legC:1.2
02:25:49.125 [MS210007] C:1.2:Answer provided. Connection(by pass mode): 63.209.144.201:26918(26919)
02:25:49.125 [MS210001] C:1.1:Answer received. RTP connection[unsecure]: 63.209.144.201:26918(26919)
02:25:49.109 Remote SDP is set for legC:1.1
02:25:48.968 Remote SDP is set for legC:1.2
02:25:47.375 [MS210007] C:1.2:Answer provided. Connection(by pass mode): 63.209.144.201:26918(26919)
02:25:47.375 [MS210001] C:1.1:Answer received. RTP connection[unsecure]: 63.209.144.201:26918(26919)
02:25:47.375 Remote SDP is set for legC:1.1
02:25:47.218 Remote SDP is set for legC:1.2
02:25:46.046 Session 116 of leg C:1.1 is confirmed
02:25:45.750 [CM503007]: Call(1): Device joined: sip:[email protected]:5100;user=phone
02:25:45.734 [CM503007]: Call(1): Device joined: sip:[email protected]:5060
02:25:45.734 [MS210007] C:1.1:Answer provided. Connection(by pass mode): public.ip:10002(10003)
02:25:45.734 [MS210001] C:1.2:Answer received. RTP connection[unsecure]: public.ip:10002(10003)
02:25:45.734 Remote SDP is set for legC:1.2
02:25:45.734 [CM503002]: Call(1): Alerting sip:[email protected]:5100;user=phone
02:25:45.656 [CM503025]: Call(1): Calling @[Dev:sip:[email protected]:5100;user=phone]
02:25:45.656 [MS210006] C:1.2:Offer provided. Connection(by pass mode): 63.209.144.201:26918(26919)
02:25:45.625 [CM503004]: Call(1): Route 1: @[Dev:sip:[email protected]:5100;user=phone]
02:25:45.625 [CM503010]: Making route(s) to <sip:[email protected]:5060;user=fax>
02:25:45.625 [MS210000] C:1.1:Offer received. RTP connection: 63.209.144.201:26918(26919)
02:25:45.625 Remote SDP is set for legC:1.1
02:25:45.609 [CM505003]: Provider:[Skype SIP - 2CH In handler] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [SipGW 0.3.19] PBX contact: [sip:[email protected]:5060]
02:25:45.609 [CM503001]: Call(1): Incoming call from 16465332025@(Ln.10001@Skype SIP - 2CH In handler) to <sip:[email protected]:5060;user=fax>
02:25:45.421 [CM503012]: Inbound out-of-office hours rule (12014674775) for 10001 forwards to Fax:888
02:25:45.421 Looking for inbound target: called=12014674775; caller=16465332025
02:25:45.421 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060;rinstance=20cd16af2d3b54ea;transport=udp SIP/2.0
Via: SIP/2.0/UDP 63.209.144.201:5060;branch=z9hG4bK-60596-4c68d9e3-e66479d5-5aedbe33
Max-Forwards: 30
Contact: <sip:[email protected]:5060;transport=udp>
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=c990d13f-13c4-4c68d9e3-e66479d5-76b781a2
Call-ID: CXC-56-67081830-c990d13f-13c4-4c68d9e3-e66479d5-6489c584
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
User-Agent: SipGW 0.3.19
Content-Length: 0

02:25:32.328 Currently active calls [none]
 
Skype does not support fax, especially T38 faxing which is what the 3cx fax server uses.

If you want to receive faxes using the 3cx fax server then you need a VOIP provider that supports it. We use Broadvox for us and our customers.
 
Ok, Does Xeloq Support it?
 
i there a way to test internaly if the fax service in working on the server. Like dailing the extention of the fax server to hear a tone?
 
djboxny said:
i there a way to test internaly if the fax service in working on the server. Like dailing the extention of the fax server to hear a tone?
Short answer - no.

Faxing over IP (FOIP) is a different protocol to VOIP.

Basically when the call comes in to the 3cx server and the 3cx knows it is to go to a fax (T38) it sends a reinvite message back saying "contact me again on this number using the t38 protocol". As sip phones do not know what t38 is then you cannot dial the extension internally.

The fax server does work. We have it on one real estate office where they have 24 DIDs for fax - one for each agent. If the fax server has a problem then it will show in the windows event logs if the service is a problem or the 3cx logs if it is a configuration problem. But other than the few well documented changes which take about 10 minutes to set up - it works out of the box.

I do not believe xeloq uses t38. We successfully use Broadvox for our sip trunks and they use t38. I will be happy to give you some contact information and setup.
 
I Contacted Broadvox already and i am getting their services. How do i connect my Laptop to the fax server to send faxes. The turorial here seems outdated.
 
can someone please tell me Step by step how to install and configure Faxing on windows server 2003 so other computers on the network can use it. I already have a provider that supports it. I cant figure it out even tried this
http://www.3cx.com/blog/docs/microsoft-fax-2003/ to no avail. Please help
 
djboxny said:
can someone please tell me Step by step how to install and configure Faxing on windows server 2003 so other computers on the network can use it. I already have a provider that supports it. I cant figure it out even tried this
http://www.3cx.com/blog/docs/microsoft-fax-2003/ to no avail. Please help

3CX does not support this any longer. There used to be a fax sending module in an older version but that is no longer the case. There are rumors that they may be a new one in the future, but I doubt this is high on their priority list.

To send a fax using Windows Fax Server, you will need a card in the server to connect to a PSTN network.
 
Ok i just got broadvox. i tried sending a fax to it from a analog fax machine connected in my house and it dint work.

SMTP server is working correctly. DID is routing to fax.

this is the logs

22:53:11.495 [CM503008]: Call(26): Call is terminated
22:53:09.589 [CM503007]: Call(26): Device joined: sip:[email protected]:5100;user=phone
22:53:09.589 [CM503007]: Call(26): Device joined: sip:[email protected]:5060
22:53:09.573 [CM503002]: Call(26): Alerting sip:[email protected]:5100;user=phone
22:53:09.495 [CM503025]: Call(26): Calling @[Dev:sip:[email protected]:5100;user=phone]
22:53:09.448 [CM503004]: Call(26): Route 1: @[Dev:sip:[email protected]:5100;user=phone]
22:53:09.448 [CM503010]: Making route(s) to <sip:[email protected]:5060;user=fax>
22:53:09.417 [CM505003]: Provider:[Broadvox GO! SIP] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Broadvox Fusion] PBX contact: [sip:[email protected]:5060]
22:53:09.402 [CM503001]: Call(26): Incoming call from 2017199243@(Ln.10002@Broadvox GO! SIP) to <sip:[email protected]:5060;user=fax>
22:53:09.355 [CM503012]: Inbound out-of-office hours rule (Broadvox Fax Line) for 10002 forwards to Fax:888
22:37:16.152 [CM504001]: Ext.100: new contact is registered. Contact(s): [sip:[email protected]:58118;rinstance=29CF3CC1/100,sip:[email protected]:11396;rinstance=6536BD0F099594ABEE3E/100]
22:37:13.558 [CM504001]: Ext.100: new contact is registered. Contact(s): [sip:[email protected]:11396;rinstance=6536BD0F099594ABEE3E/100]
22:36:44.370 [CM504001]: Ext.101: new contact is registered. Contact(s): [sip:[email protected]:49651;rinstance=5adfb23a56ac6441/101]
21:48:55.183 [CM504002]: Ext.101: a contact is unregistered. Contact(s): []
21:43:01.933 [CM504004]: Registration succeeded for: 10002@Broadvox GO! SIP
21:43:01.652 [CM504003]: Sent registration request for 10002@Broadvox GO! SIP
21:42:10.027 [CM504007]: Next attempt to register 10002@Broadvox GO! SIP is scheduled in 5 minute
21:42:10.011 [CM504005]: Registration failed for: 10002@Broadvox GO! SIP; Cause: 403 Forbidden; from IP:208.93.224.229:5060
21:42:09.683 [CM504003]: Sent registration request for 10002@Broadvox GO! SIP
21:41:07.777 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone
21:41:05.917 [CM504010]: Fax Service: unregistered contact sip:
21:39:51.777 [CM503008]: Call(25): Call is terminated
21:39:49.855 [CM503007]: Call(25): Device joined: sip:[email protected]:5100;user=phone
21:39:49.855 [CM503007]: Call(25): Device joined: sip:[email protected]:5060
21:39:49.839 [CM503002]: Call(25): Alerting sip:[email protected]:5100;user=phone
21:39:49.745 [CM503025]: Call(25): Calling @[Dev:sip:[email protected]:5100;user=phone]
21:39:49.683 [CM503004]: Call(25): Route 1: @[Dev:sip:[email protected]:5100;user=phone]
21:39:49.683 [CM503010]: Making route(s) to <sip:[email protected]:5060;user=fax>
21:39:49.683 [CM505003]: Provider:[Broadvox GO! SIP] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Broadvox Fusion] PBX contact: [sip:[email protected]:5060]
21:39:49.683 [CM503001]: Call(25): Incoming call from 6465332025@(Ln.10002@Broadvox GO! SIP) to <sip:[email protected]:5060;user=fax>
21:39:49.652 [CM503012]: Inbound out-of-office hours rule (Broadvox Fax Line) for 10002 forwards to Fax:888
 
i just spoke to Broadvox. They sampled a fax call then told me that 3CX Sends t38 ring invite, broadvox sends 200 ok in t38 codec and as soon as the they send the 200 ok 3cx terminates the call. Y is 3CX not alowwing the connection to finish.????
 
Have you set up everything in 3cx server according to the instructions?

Did you change the two templates? Do you have a static public ip for the 3cx?

Can you take a wireshark capture so we can see where it is going wrong?
 
Yes i have a Static IP, with No firewall at all. here is the verbose logs

10:55:09.230 [CM503008]: Call(2): Call is terminated
10:55:09.105 Remote SDP is set for legC:2.1
10:55:08.698 Remote SDP is set for legC:2.2
10:55:07.355 Session 159 of leg C:2.1 is confirmed
10:55:07.214 [CM503007]: Call(2): Device joined: sip:[email protected]:5100;user=phone
10:55:07.214 [CM503007]: Call(2): Device joined: sip:[email protected]:5060
10:55:07.214 [MS210007] C:2.1:Answer provided. Connection(by pass mode): 192.168.1.145:10002(10003)
10:55:07.214 [MS210001] C:2.2:Answer received. RTP connection[unsecure]: 192.168.1.145:10002(10003)
10:55:07.214 Remote SDP is set for legC:2.2
10:55:07.214 [CM503002]: Call(2): Alerting sip:[email protected]:5100;user=phone
10:55:07.167 [CM503025]: Call(2): Calling @[Dev:sip:[email protected]:5100;user=phone]
10:55:07.167 [MS210006] C:2.2:Offer provided. Connection(by pass mode): 64.158.162.71:24844(24845)
10:55:07.120 [CM503004]: Call(2): Route 1: @[Dev:sip:[email protected]:5100;user=phone]
10:55:07.120 [CM503010]: Making route(s) to <sip:[email protected]:5060;user=fax>
10:55:07.120 [MS210000] C:2.1:Offer received. RTP connection: 64.158.162.71:24844(24845)
10:55:07.120 Remote SDP is set for legC:2.1
10:55:07.120 [CM505003]: Provider:[Broadvox GO! SIP] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Broadvox Fusion] PBX contact: [sip:[email protected]:5060]
10:55:07.120 [CM503001]: Call(2): Incoming call from 2017199243@(Ln.10002@Broadvox GO! SIP) to <sip:[email protected]:5060;user=fax>
10:55:07.105 [CM503012]: Inbound out-of-office hours rule (Broadvox Fax Line) for 10002 forwards to Fax:888
10:55:07.105 Looking for inbound target: called=9738704381; caller=2017199243
10:55:07.089 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 208.93.224.229;rport=5060;branch=z9hG4bKaB1yBt5KFrQ0j
Max-Forwards: 69
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: "2017199243"<sip:[email protected]>;tag=62gBmyvtyX4FQ
Call-ID: e22ae6e1-2f88-122e-eda0-a4badb456be6
CSeq: 1288234 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
User-Agent: Broadvox Fusion
Allow-Events: talk, hold, refer
Privacy: none
P-Asserted-Identity: "2017199243" <sip:[email protected]>
Content-Length: 0
X-FS-Support: update_display

10:54:55.277 Currently active calls [none]
 
Those logs dont help as we need to see the sip information being sent and received.

Before we go into that, can you confirm you have set up the 3cx fax in accordance with the instructions here

http://www.3cx.com/blog/docs/fax-nat-traversal/
 
i just did that, does that require all systems restarted?
 
this is the wiresharck capture of a 10 packet fax call trying to come on. I have in the fax server settings to bind to the Static IP is that correct.

No. Time Source Destination Protocol Info
88397 93.234660 208.93.226.213 public.ip SIP/SDP Request: INVITE sip:[email protected]:5060, with session description

Frame 88397: 1066 bytes on wire (8528 bits), 1066 bytes captured (8528 bits)
Ethernet II, Src: Cisco_13:82:b8 (a4:0c:c3:13:82:b8), Dst: HewlettP_6d:22:6e (00:0f:20:6d:22:6e)
Internet Protocol, Src: 208.93.226.213 (208.93.226.213), Dst: public.ip (public.ip)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No. Time Source Destination Protocol Info
88445 93.360389 public.ip 208.93.226.213 SIP Status: 100 Trying

Frame 88445: 337 bytes on wire (2696 bits), 337 bytes captured (2696 bits)
Ethernet II, Src: HewlettP_6d:22:6e (00:0f:20:6d:22:6e), Dst: Cisco_13:82:b8 (a4:0c:c3:13:82:b8)
Internet Protocol, Src: public.ip (public.ip), Dst: 208.93.226.213 (208.93.226.213)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No. Time Source Destination Protocol Info
88513 93.472774 public.ip 208.93.226.213 SIP/SDP Status: 200 OK, with session description

Frame 88513: 766 bytes on wire (6128 bits), 766 bytes captured (6128 bits)
Ethernet II, Src: HewlettP_6d:22:6e (00:0f:20:6d:22:6e), Dst: Cisco_13:82:b8 (a4:0c:c3:13:82:b8)
Internet Protocol, Src: public.ip (public.ip), Dst: 208.93.226.213 (208.93.226.213)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No. Time Source Destination Protocol Info
88705 93.586620 208.93.226.213 public.ip SIP Request: ACK sip:[email protected]:5060

Frame 88705: 436 bytes on wire (3488 bits), 436 bytes captured (3488 bits)
Ethernet II, Src: Cisco_13:82:b8 (a4:0c:c3:13:82:b8), Dst: HewlettP_6d:22:6e (00:0f:20:6d:22:6e)
Internet Protocol, Src: 208.93.226.213 (208.93.226.213), Dst: public.ip (public.ip)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No. Time Source Destination Protocol Info
88718 93.626878 64.152.60.71 public.ip RTP PT=ITU-T G.711 PCMU, SSRC=0xD1753E18, Seq=0, Time=0, Mark

Frame 88718: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits)
Ethernet II, Src: Cisco_13:82:b8 (a4:0c:c3:13:82:b8), Dst: HewlettP_6d:22:6e (00:0f:20:6d:22:6e)
Internet Protocol, Src: 64.152.60.71 (64.152.60.71), Dst: public.ip (public.ip)
User Datagram Protocol, Src Port: 25138 (25138), Dst Port: documentum (10002)
Real-Time Transport Protocol

No. Time Source Destination Protocol Info
90049 94.967050 public.ip 208.93.226.213 SIP/SDP Request: INVITE sip:[email protected]:5060, in-dialog, with session description

Frame 90049: 955 bytes on wire (7640 bits), 955 bytes captured (7640 bits)
Ethernet II, Src: HewlettP_6d:22:6e (00:0f:20:6d:22:6e), Dst: Cisco_13:82:b8 (a4:0c:c3:13:82:b8)
Internet Protocol, Src: public.ip (public.ip), Dst: 208.93.226.213 (208.93.226.213)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No. Time Source Destination Protocol Info
90154 95.023679 208.93.226.213 public.ip SIP Status: 100 Trying

Frame 90154: 402 bytes on wire (3216 bits), 402 bytes captured (3216 bits)
Ethernet II, Src: Cisco_13:82:b8 (a4:0c:c3:13:82:b8), Dst: HewlettP_6d:22:6e (00:0f:20:6d:22:6e)
Internet Protocol, Src: 208.93.226.213 (208.93.226.213), Dst: public.ip (public.ip)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No. Time Source Destination Protocol Info
90371 95.250190 208.93.226.213 public.ip SIP/SDP Status: 200 OK, with session description

Frame 90371: 961 bytes on wire (7688 bits), 961 bytes captured (7688 bits)
Ethernet II, Src: Cisco_13:82:b8 (a4:0c:c3:13:82:b8), Dst: HewlettP_6d:22:6e (00:0f:20:6d:22:6e)
Internet Protocol, Src: 208.93.226.213 (208.93.226.213), Dst: public.ip (public.ip)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No. Time Source Destination Protocol Info
90407 95.354827 public.ip 208.93.226.213 SIP Request: ACK sip:[email protected]:5060

Frame 90407: 498 bytes on wire (3984 bits), 498 bytes captured (3984 bits)
Ethernet II, Src: HewlettP_6d:22:6e (00:0f:20:6d:22:6e), Dst: Cisco_13:82:b8 (a4:0c:c3:13:82:b8)
Internet Protocol, Src: public.ip (public.ip), Dst: 208.93.226.213 (208.93.226.213)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No. Time Source Destination Protocol Info
90431 95.467993 public.ip 208.93.226.213 SIP Request: BYE sip:[email protected]:5060

Frame 90431: 498 bytes on wire (3984 bits), 498 bytes captured (3984 bits)
Ethernet II, Src: HewlettP_6d:22:6e (00:0f:20:6d:22:6e), Dst: Cisco_13:82:b8 (a4:0c:c3:13:82:b8)
Internet Protocol, Src: public.ip (public.ip), Dst: 208.93.226.213 (208.93.226.213)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol

No. Time Source Destination Protocol Info
90444 95.527015 208.93.226.213 public.ip SIP Status: 200 OK

Frame 90444: 533 bytes on wire (4264 bits), 533 bytes captured (4264 bits)
Ethernet II, Src: Cisco_13:82:b8 (a4:0c:c3:13:82:b8), Dst: HewlettP_6d:22:6e (00:0f:20:6d:22:6e)
Internet Protocol, Src: 208.93.226.213 (208.93.226.213), Dst: public.ip (public.ip)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
 
ok this is a Update,

in the fax server setting i have set the network interface to the local IP of the server instead of the Static one.

Now if i dial the fax server extention from a internal Extention i hear the Fax tones and sounds. wich means it is working. But anyone dialing into the fax DID gets the same disconnection.
 
UPDATEE!!!


I have succesfully received a fax BUT only thru a grandstream handytone 502 and a analog fax machine.

here is the setup

Broadvox VOIP --> 3cx --> Handytone ATA 502 (t38 pass thru mode) --> Analog Fax machine.

I worked great. But what i really want is the 3cx Fax server to work.
 
You need to restart the fax server to make the changes.

I dont see any t38 requests coming in or out.

Restart 3cx if you can and then run wireshark and see if you can get any t38 requests.

EDIT: I missed P2 of the thread with the latest updates from you.

In the fax template configs should make NAT support true and uncomment the sip port 5100 (make sure open on firewall too).
Then in the other template you should need to make the kind of t38 transport UDP and then put the local ip address as your internal ip and the (nat) local media ip address as your public static ip address.

Then go into the advance settings , custom parameters and make Faxdirectsdp set to 1.

Then restart the 3cx and it should all work.
 
OK, According to Broadvox wireshark capture this is whats happening.

a Fax comes into the fax line did then it is sent to the 3cx fax server, the fax server requests port 10002 so broadvox acknowledges it and tries to send the media thru that port. BUT according to broadvox port 10002 is unreachable here is the logs

1 0.000000 208.93.224.229 my.public.ip SIP/SDP Request: INVITE sip:[email protected]:5060, with session description
2 0.161303 my.public.ip 208.93.224.229 SIP Status: 100 Trying
3 0.267339 my.public.ip 208.93.224.229 SIP/SDP Status: 200 OK, with session description
4 0.268241 208.93.224.229 my.public.ip SIP Request: ACK sip:[email protected]:5060
5 0.387340 208.93.224.229 my.public.ip RTP PT=ITU-T G.711 PCMU, SSRC=0x46BDD664, Seq=41360, Time=39944
6 0.427193 208.93.224.229 my.public.ip RTP PT=ITU-T G.711 PCMU, SSRC=0x46BDD664, Seq=41361, Time=40264
7 0.447356 my.public.ip 208.93.224.229 ICMP Destination unreachable (Port unreachable)
8 0.447424 208.93.224.229 my.public.ip RTP PT=ITU-T G.711 PCMU, SSRC=0x46BDD664, Seq=41362, Time=40424
9 0.467272 208.93.224.229 my.public.ip RTP PT=ITU-T G.711 PCMU, SSRC=0x46BDD664, Seq=41363, Time=40584
10 0.480571 my.public.ip 208.93.224.229 ICMP Destination unreachable (Port unreachable)
11 0.487397 208.93.224.229 my.public.ip RTP PT=ITU-T G.711 PCMU, SSRC=0x46BDD664, Seq=41364, Time=40744
12 0.500568 my.public.ip 208.93.224.229 ICMP Destination unreachable (Port unreachable)

and after many tries they get this

140 1.769070 my.public.ip 208.93.224.229 SIP/SDP Request: INVITE sip:[email protected]:5060, in-dialog, with session description
141 1.769578 208.93.224.229 my.public.ip SIP Status: 100 Trying
142 1.787036 my.public.ip 208.93.224.229 ICMP Destination unreachable (Port unreachable)
145 41.865898 my.public.ip 208.93.224.229 SIP Request: BYE sip:[email protected]:5060
146 41.867084 208.93.224.229 my.public.ip SIP Status: 200 OK
147 41.867199 208.93.224.229 my.public.ip SIP Status: 487 Session Terminated
148 41.925557 my.public.ip 208.93.224.229 SIP Request: ACK sip:[email protected]:5060

media port info audio 10002 RTP/AVP 0
 
Do you have those ports open and forwarded in your router/firewall?

Why is it 10002? It should be 10000. Did you change it in the 3CX in_session.cfg? I dont know if this makes a difference or not.

But either you have the wrong thing being sent from the config or the port is not open on your end.
 
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