Trouble with inbound on freepbx

I have two numbers
1) 0844986xxxx (in 'sip account')
2) 0844484xxxx (under 'my numbers')

I am trying to setup both to come to my asterisk+freepbx.

My account is registered fine and I can make outgoing calls (tested echo test and dialled a uk toll free number)

However I can not get inbound to work.
Since I donot have way to call from PSTN to test the inbound, I created another voipuser account and tried to call from that account to this account by directly dialling both the numbers (staring with 08xxx). However here is what I get ..

1) With number 1 (sip account) it keeps on ringing (i do not see any incoming trace in asterisk)
It gets stuck at
-------------------------
Executing Dial("SIP/2001-b7b00968", "SIP/voipuser2/0844986xxxx|120|mW") in new stack
-- Called voipuser2/0844986xxxx
-- Started music on hold, class 'default', on SIP/2001-b7b00968
-------------------------

2) With second number, I set it to forward using SIP to
0844986xxxx [!at] mydyn.dns.name.com (replace the [!at] with a @)
When I call this number it just gives congestion.
------------------
Called voipuser/0844484xxxx
-- Started music on hold, class 'default', on SIP/2001-b7b2d6d8
-- SIP/voipuser-09cd6078 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Stopped music on hold on SIP/2001-b7b2d6d8
-- Executing Goto("SIP/2001-b7b2d6d8", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing NoOp("SIP/2001-b7b2d6d8", "Dial failed due to CONGESTION") in new stack
-- Executing Macro("SIP/2001-b7b2d6d8", "outisbusy|") in new stack
-- Executing Playback("SIP/2001-b7b2d6d8", "all-circuits-busy-now|noanswer") in new stack
-- Playing 'all-circuits-busy-now' (language 'en')
-- Executing Playback("SIP/2001-b7b2d6d8", "pls-try-call-later|noanswer") in new stack
-- Playing 'pls-try-call-later' (language 'en')
--------------------------------------------------

I have other services like FWD, Some commerical voip provider, freedigits etc configured and all of them working fine for incoming and outgoing so it is not nat or any basic asterisk issue.

Please help. Let me know if I can provide any more information.

I find it surprising that when people just say that 'my service is not working' there is almost immediate response saying that one needs to provide more information. I posted most of the information upfront and now there is no response.

Is this likely to be an Asterisk issue?

Where are you test dialling the numbers from?

Quote:

there is no response.

We don't do this for a living I'm afraid. VoIP User is non-profit and no-one here gets paid anything. We do our best to support users but it is what it is - quality of support often depends on who's around and who has time

If you need a support team you know you can get in a certain time frame then you need a commercial provider.

Dean

dean :

Is this likely to be an Asterisk issue? Where are you test dialling the numbers from? Quote: there is no response. We don't do this for a living I'm afraid. VoIP User is non-profit and no-one here gets paid anything. We do our best to support users but it is what it is - quality of support often depends on who's around and who has time If you need a support team you know you can get in a certain time frame then you need a commercial provider. Dean

I understand and apologize. In my frustation ( have been trying to get it to work for last several days ) I posted that message. No hard feelings.

It must be my asterisk issue, since all other people are able to get incoming calls. I am not able to pin point where though. I am able to make 'echo' calls so outgoing is fine. However for incoming not sure what I am doing wrong. To test incoming I am using another voipuser account. I tried dialling
0448xxxx
and
08xxx
Both of them did not work. Dialling 0448xxx doesn't comlete the call (log above) and dialling 08xxx simply comes back as congestion.

To help kill a few streams of problems.

When a call comes in from VoipUser it will be in international format i.e. 44844xxx, so you will need to configure FreePBX to recognise that in Inbound routing

Once done (depending on how the extensions.conf is written) you should find that dialing your Voipuser number through the PBX will result in the PBX performing the default reaction for stupidity i.e. dialing yourself.

In the second case, unless you have set up a SIP context called 0844484xxxx, then the PBX will not respond.
There is also a small glitch in My Numbers settings, what you configure in Destination 1 Options is not necessarily called Destination 1 for Termination. (Try selecting Destination 2 instead)

It is considerably easier to configure if you use an external connection to test, it needn't cost anything, you won't be billed until the PBX answers the call

rgower :

1) When a call comes in from VoipUser it will be in international format i.e. 44844xxx, so you will need to configure FreePBX to recognise that in Inbound routing 2) Once done (depending on how the extensions.conf is written) you should find that dialing your Voipuser number through the PBX will result in the PBX performing the default reaction for stupidity i.e. dialing yourself. 3) It is considerably easier to configure if you use an external connection to test, it needn't cost anything, you won't be billed until the PBX answers the call

for 1. yes that is taken care of
for 2, I created another voipuser account and verified that it was being used to call the first one. (Maybe that is the problem) So asterisk was using 1 voipuser trunk to call another.

3) I will try that when I get back home.

However didn't you say that one voipuser account should be able to call another voip user account (using 08xxx format) and they get routed internally, rather than on PSTN. However to asterisk it should appear no differently. Right ?

guess what.. there was NOTHING wrong with my setup.. destination 2 tip worked..
i tried calling from skype and it worked !
now I will try to get as much incoming calls as possible once I get my asterisk custom context start working