Sip Trunk connections problem.

[toolfolks]

Hi guys & girls.
From a background of windows / mac database 'stuff' I've taken the plunge into Linux. Built a Debian 10 server box ( all went well ) then installed FusionPBX ( all went well after lots of typing of commands ).
I can make internal calls on extensions and mobile sip phones.

However our Sip provider has supplied the connection details and added our IP at their end but I'm unable to connect.
Unfortunately this was late Friday so not been able to get through to their technical support.

I have added the following :
Username = xxxxxxx
Password = X6-xxxxxx
Domain = sip.xxxx.com

Connection to event socket failed.

I've tried zillions of different settings but get the same result ( I also tried another free Sip Trunk trial ans get the same issue ).

So I'm hoping I need to maybe open ports ?
Set up a firewall ?

Any help appreciated at idiot level please ..........

 

[nktech1135]

Who is your sip carrier?
Do they require registration? or is it done by some other method?
You say you added user name, password, and domain, but you didn't say where. Was it in accounts>gateways? or somewhere else?

Sorry for answering questions with questions, but sometimes that's the best way to get answers.

 

[toolfolks]

No problem....

just need to get this working so I can add some more pieces to the application.

 

[toolfolks]

Sip provider is xinix
They state they have added my IP
They sent

Username = xxxxxxx
Password = X6-xxxxxx
Domain = sip.xinix.com

 

[Adrian Fretwell]

You will need to get some more diagnostic information. What do the FreeSwitch logs say? Have you tried getting a packet capture to see what messages are being exchanged between your box and your provider?

 

[toolfolks]

Hi Adrian,
new to this.
1. Where are the Freeswitch Logs stored ?
2. How do I get a packet capture ?

Cheers

 

[Adrian Fretwell]

Hi, FreeSwitch logs are in /var/log/freeswitch. You can watch the log live with the tail command and the -f switch (CTRL-C to end).

tail -f /var/log/freeswitch/freeswitch.log

I mention how I do packet captures in this thread:
https://www.pbxforums.com/threads/inbound-call-dropping.4038/#post-14765

 

[toolfolks]

Hi,

logged into the debian box as root.
There is no freeswitch.log file only freeswitch.xml.fsxml and folder xml_cdr

I have followed the instructions on packet capture and added sip.xininxworld.com to the IP list and log filename xinixlog.txt

I stopped and started the gateway.
I went to status and clicked sip status.

I checked xinixlog.txt but the file is empty.

 

[ewdpb]

@toolfolks , log in to the Linux server console. Type fs_cli. What do you see? Try restarting the GW, what do you see? Did you add the ACL rules to allow connections from your provider? Follow this.

 

[Adrian Fretwell]

@toolfolks , log in to the Linux server console. Type fs_cli. What do you see? Try restarting the GW, what do you see? Did you add the ACL rules to allow connections from your provider? Follow this.

I have assertained that the box is on a private LAN, I don't yet know if it is even talking to the outside world - one step at a time...

 

[toolfolks]

I am trying to open the ports as per docs here

https://docs.fusionpbx.com/en/latest/firewall.html

I am on TalkTalk in the UK.

If I try to open ports 5060 - 5090
I get the error:

The range of external port cannot include 514,68,53,546,1701,8443,8080,5060,28090,50000-50020.

I get the same when trying 16384-32768

In the router there is a setting AGL


ALG


What's this?
The application layer gateway parses application layer information to ensure communication over the application layer.

• Enable PPTP ALG:
  The Point to Point Tunneling Protocol (PPTP) is mainly used in virtual private networks (VPNs) to enable Internet users to securely access enterprise VPNs.
• Enable H.323 ALG:
  H.323 transmits data of video conferencing and multimedia conferencing services.
• Enable RTSP ALG:
  The Real Time Streaming Protocol (RSTP) transmits multimedia data.
• Enable SIP ALG:
  The Session Initiation Protocol (SIP) transmits data of VoIP services.
• SIP port:
  This parameter specifies the port number of the SIP proxy server. The SIP proxy server can use this port number to connect to the network. For details, consult the SIP proxy platform carrier.

These are all ticked and sip port is 5060.
Enable PPTP ALG:

Enable H.323 ALG:

Enable RTSP ALG:

Enable SIP ALG:

SIP port:

If I check for open ports 5060 is not open ???

Does this mean the ALG setting allows the Sip traffic through ????

Out of my depth here guys.

 

[Adrian Fretwell]

I always advise people to disable SIP ALG. What make of router do you have?

 

[toolfolks]

HG633

 

[Adrian Fretwell]

HG633

Not a router I am familiar with. Disble all the ALGs and then see if it will let you open the ports.

 

[toolfolks]

I disabled all the ALG
same error
The range of external port cannot include 514,68,53,546,1701,8443,8080,5060,28090,50000-50020.

 

[Adrian Fretwell]

Your router is not helping you. Many of the cheap provider supplied routers have limitations. There are a number of options open to you:

1. Configure FusionPBX so it does not use ports that your router does not allow.
2. Get a different router (If yours can be configured into modem mode, then a Mikrotik is a good choice).
3. Get a cheap VM with a public IP address and play with FusionPBX on there.

Option 3 will eliminate any potential NAT issues that may otherwise trip you up.

Also, just as an aside, ealier in your post you mentioned sip.xininxworld.com, For me, this does not resolve to an IP address, in other words it does not appear to be valid.

 

[toolfolks]

It’s Sip.Xinixworld.com whoops

If I change the ports setting to say 5000 will my service provider have to change settings at their end ?

 

[Adrian Fretwell]

If your gateway connects by registration, then no. Otherwise the provider will need to know what port to send to. In the case of the default SIP external profile it is port 5080. (5060 for the Internal profile).

 

[toolfolks]

I've made progress.........

I changed the external ports in FusionPBX to 5000 & 5010
I opened these ports on the router and forwarded to 5060 & 5061

Just made a call to our 0333 number and a phone rang.........

3 days later....

Thanks for the input so far Adrian.

Now world wide domination......

 

[toolfolks]

Worldwide domination on hold......
I now cannot connect any GSwave apps.
I'm using the GSWave application bar code method but they don't connect

user 101
pass 101
SipServer test.toolfolks.com

Is this related to changing the ports ??????