Please please could some one help me,
I have been struggling the whole of the last week to try and get a clients sip trunk working.
It is receiving calls happily, but the outgoing calls seem to come and go weirdly enough. They work some times and then it stops working for a while.
It gives me the congestion message?
I only have one sip trunk (the one i’m trying to set-up now) which support multiple concurrent calls, and the sip provider is saying that ti must be my configuration, and they don’t provide support for 3rd party applications.
Please please could someone help me out.
Here is my config for the trunk:
I have added the following to my SIP configuration:
and I have configured the NAT, but as far as I understand that only applies for the incoming?
NAT is as follows:
IP Config: Dynamic IP
Dynamic host: blah.dyndns.org
local networks: 192.168.1.0 / 255.255.255.0
What am I missing here?
From their 3CX setup guide:
User: [You VoIP number, eg. 27870010000]
ID: [You VoIP number, eg. 27870010000]
Password: [Your VoIP password as provided by Nexus]
Use As: [SIP Account]
Local PBX IP: 22.214.171.124
External PBX IP: 126.96.36.199
PBX Port: 5060
Record Calls [off]
Phonebook “+” to: 00
Phonebook prefix: [blank]
Local SIP port: 5065
First RTP port: 4000
TCP transport: [off]
NAT Helper: [on]
Echo Cancellation: [on]
G.711 (alaw): [on]
G.711 (ulaw): [on]
So you will want to match this in your settings.
Thank you very much for your response and especially your effort in locating their 3CX guide, I have looked at that before when trying to find settings. Do my settings (listed in initial post) not respond to theirs? I am using freepbx to try and connect the trunk (if I never stated that previously)
The main concern I am having is that, the trunk is registering and connecting like everything is fine, Just when I call its giving me the congestion message? As I don’t really know the internals of freepbx, What would cause the trunk to register, connect etc. but then on outgoing calls it gives me the congestion message. Which to me means that in the outbound route, the sip trunk is not offering up its available channels, And the outbound route says oh I cant handle this message?
After much fighting with nexus they gave me this: (which is what I have based my initial config on.)
In general however you sip.conf file should look something to the effect of below:
register => 2787001XXXX:[email protected]
End of message,
But something some where is not hundreds
Post the trace of a failed call.
Seem as though my web interface just randomly disappeared.
Where can I find the specific log which will contain that data?
Sorry for the noob question