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foufee

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#21742 5-May-2008 19:59
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Hi,
  I'm trying to get a basic Trixbox setup working, and have followed the instructions (and double checked them), and at the moment I cannot call in or out.
The things that are "special" about my setup is that I have an IPCop firewall running, and a provisioned PAP2T with a different phone line. What are the chances that they will confuse things.
  Where can I find out if the trunk is connected to VFX correctly etc, there doesn't seem to be an easily visible log (or is /var/log/asterix/ it)
   Thanks in advance.

   Matthew

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nate
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  #128669 5-May-2008 20:18
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Can you remove IPCop from your network, or put your Trixbox in a DMZ?  This would isolate IPCop as being the issue, if it doesn't, you know something isn't quite right with your box.

tonyhughes
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  #128682 5-May-2008 21:39
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Are you trying to use VFX on the Trixbox?

Have you asked WxC to set you up as a VFX+Asterisk user?







 
 
 
 


foufee

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  #128704 6-May-2008 06:26
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I'll try setting the Trixbox up in the DMZ today, and yes to VFX, and yes to having asked WxC to set me up as an asterix user.

joshp
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  #128709 6-May-2008 08:29
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Hi Foufee,

It looks like your Asterisk box is currently registered so your registrations are coming through.  I however am not seeing any calls being generated from your number, so i suspect they aren't getting out.

Couple of things that you can do to check the flow of data.  If you use something such as ngrep -d eth0 port 5060 and host 58.28.20.150

what this command will basically do is show you what ethernet packets are going out of your eth0 interface on port 5060 and packets destined for us.

If the information is getting out of your box, then I suggest looking at your router or anything upstream from your server. If the information isn't being sent from your box then you could have some routing issues or dialplans not setup correctly.. go over your configuration files.

As I say we are not yet receiving your calls so the call will not be leaving your network.

Hope this helps.

Cheers

Josh




foufee

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  #128718 6-May-2008 08:51
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OK,
I have TrixBox in the DMZ, and have forwarded port 5060 (TCP/UDP) from my external IP on the IPCop box to the TrixBox server. Running ngrep shows the registrations are getting through, but when I try making a call from the internal network, all I get is "All circuits are busy now" .
I'm currently setting up another internal extension and seeing if I can call that.

edit:Calling internal extensions work.

nate
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  #128724 6-May-2008 09:13
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Have a read through Tony's Trixbox guide here just to check you haven't missed anything obvious.


foufee

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  #128741 6-May-2008 10:32
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I used that guide originally to set it up, but just tried it all again.

This might be a silly question, but my number is a wellington (04) number, I take it in the guide, where 9 is I replace it with 4, except for the outbound route bit?

Cheers..
Matthew

 
 
 
 


joshp
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  #128743 6-May-2008 10:38
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Hrmm well looking at your inbound calls you are sending us a not acceptable here message, what this basically means is that you do not accept any of the codecs that are being offered to you.  What codecs have you currently enabled on your end?

it should look something like:

allow=ulaw
allow=alaw
allow=g729

etc

this also dictates your codec preference, the first in the list will be chosen etc.

Cheers




foufee

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  #128746 6-May-2008 10:49
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In the trunk settings under "PEER DETAILS" it is ulaw and alaw, on X-Lite, ulaw, and alaw are there as G711 ulaw and G711 alaw.

One other thing, I'm using Trixbox 2.6.0.7

joshp
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  #128747 6-May-2008 10:52
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Something on your side is not allowing those codecs, we are offering up g711u, g711a and g729.

Are you seeing the inbound call message hit the Asterisk box? Try running Asterisk in debug mode as well to see if it gives you any detail on what is happening.  This can be done by the following command:

#asterisk -vvvvvvr

the number of v's indicates the verbosity of the output, so the more v's the more detail you will get.

Cheers

Josh




foufee

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  #128753 6-May-2008 11:29
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I've checked /usr/lib/asterix/modules, and there is are codec_ulaw,codec_alaw and a codec_g726 shared objects

I've blanked out my phone numbers with ?

I see (via ngrep), a incoming call.

filter: (ip) and ( port 5060 and host 58.28.20.150 )
#
U 58.28.20.150:5060 -> 192.168.2.1:5060
INVITE sip:483????6@192.168.2.1;transport=udp SIP/2.0..From: "Matthew Huck"
;tag=96141c3a-13c4-481f94ec-cf82a514-465721e6..T
o: "Matthew Huck"..Call-ID: BW1114530180605081404
035788@10.251.1.11..CSeq: 1 INVITE..Via: SIP/2.0/UDP 58.28.20.150:5060;bran
ch=z9hG4bK-d020-481f94ec-cf82a514-1d51b327..Supported: 100rel..Accept: mult
ipart/mixed,application/media_control+xml,application/sdp..Max-Forwards: 9.
.Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY..Contact:
p:483?????1@as.wxcnz.net:5060;maddr=58.28.20.150;transport=UDP>..Content-Typ
e: application/sdp..Content-Length: 335....v=0..o=BroadWorks 7029724 1 IN I
P4 58.28.20.150..s=-..c=IN IP4 58.28.20.150..t=0 0..m=audio 36976 RTP/AVP 1
8 0 8 100 101..a=rtpmap:18 G729/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA
/8000..a=rtpmap:100 NSE/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:100
192-193..a=fmtp:101 0-15..a=ptime:20..a=bsoft:1 image udptl t38..a=sendrec
v..
#
U 192.168.2.1:5060 -> 58.28.20.150:5060
SIP/2.0 488 Not acceptable here..Via: SIP/2.0/UDP 58.28.20.150:5060;branch=
z9hG4bK-d020-481f94ec-cf82a514-1d51b327;received=58.28.20.150..From: "Matth
ew Huck";tag=96141c3a-13c4-481f94ec-cf82a514-465
721e6..To: "Matthew Huck";tag=as2d270683..Call-ID
: BW1114530180605081404035788@10.251.1.11..CSeq: 1 INVITE..User-Agent: Aste
risk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIF
Y..Supported: replaces..Content-Length: 0....
#
U 58.28.20.150:5060 -> 192.168.2.1:5060
ACK sip:48311266@192.168.2.1;transport=udp SIP/2.0..From: "Matthew Huck"
p:48311161@as.wxcnz.net>;tag=96141c3a-13c4-481f94ec-cf82a514-465721e6..To:
"Matthew Huck";tag=as2d270683..Call-ID: BW1114530
180605081404035788@10.251.1.11..CSeq: 1 ACK..Via: SIP/2.0/UDP 58.28.20.150:
5060;branch=z9hG4bK-d020-481f94ec-cf82a514-1d51b327..Max-Forwards: 69..Cont
act: ..Con
tent-Length: 0....
exit
6 received, 0 dropped



As for outgoing calls, the only thing I notice in the astertix debug is:
macro dialout-trunk-predial-hook
gotoif bypass|1
gotoif 0?customtrunk
dial ("SIP/100-0a16e520","SIP/VFX SIP Trunk/83????1|300|"
Couldn't call VFX SIP Trunk/83????1

joshp
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  #128757 6-May-2008 11:40
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if ngrep is not showing up anything for your outbound calls then asterisk is unable to process the call, check back through tony's guide to ensure that you have the correct routing rules setup, also i note in his guide that he is using 9 to initiate the outbound call, are you dialing 9 and then the number that you wish to dial?

Cheers

Josh




foufee

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  #128784 6-May-2008 13:01
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Yay, all working...there is a mistake in the guide...

The the PEER details it has

allow=ulaw&alaw
...
...
disallow=all

This means that no codec is allowed.

Move disallow=all to the top, and it all works...well, I can call in and out, but inbound I don't hear the other party, and they don't hear me.  Outband works fine.

joshp
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  #128787 6-May-2008 13:15
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Excellent.. Good to hear..




sbiddle
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  #128794 6-May-2008 13:45
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foufee: Yay, all working...there is a mistake in the guide...

The the PEER details it has

allow=ulaw&alaw
...
...
disallow=all

This means that no codec is allowed.

Move disallow=all to the top, and it all works...well, I can call in and out, but inbound I don't hear the other party, and they don't hear me.  Outband works fine.


The order makes no difference. Are you sure there was nothing else you changed?

Good to see that you now have it working though!

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