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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: Drifting on August 18, 2011, 12:22:41 PM
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I have been having rather a nightmare configuring Sark with Voipon, neither are the problems Sark's or Voipon. It's me not having a lot of idea as to what I should be entering for an IAX trunk when Voipon are not listed as a trunk option.
Seem to have the trunk alive, but so far will not accept an incoming call.
Version: sail-2.2.4-54
Regards Paul
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Hi Paul
Can you post the Voipon settings please? Usually, with other IAX trunk vendors, you need to log in to a portal and point the trunk at your PBX IP (Gradwell and Telappliant both work this way). Have you done this for this trunk?
Kind Regards
Jeff
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Yes, I have set the options on the voipon service. And have them confirmed by their excellent support people. It is pointing to my IP.
Perhaps this might help in explaining what they require to be set? :-
https://www.voipon.co.uk/helpdesk/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=103&nav=0,1,10
Regards Paul
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Perhaps this might help in explaining what they require to be set? :-
It's not very clear from their documentation but it looks as though you need to create a DDI (PTT_DiD_Group in SARK) for the dialled number (your voipon number) and use that to route your call.
If that doesn't work and they can't tell you what they are sending then you will need to run a trace either from wireshark or tcpdump. You can also run a trace from the asterisk console by doing
iax2 set debug on
and then sending a call in and seeing what you get.
Best
S
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In for a penny, so might as well make myself look a total dunce. I have no idea how to do what you suggested, that was apart from the debug part and wireshark. So in essence your saying go read the sark documentation? unless your feeling kind and explain the DDI part a little more? like where do you create such a thing?
Paul
Ever the pain.
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in trunks; create a new trunk and choose PTT_DiD_Group from the drop down. The rest is pretty much like creating any other trunk. V2.x reference page for DiD is here
http://www.sailpbx.com/twiki/bin/view/Main/DocChapter095
Best
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Well I read the document link you sent, it is rather terse.
Still not sure exactly what I am supposed to be putting where.
I think my problems are a little more basic, and I am under the impression it is the IAX truck, and the settings for the account. If you are confused by their documentation, then what hope have I?
I did the debug on IAX and saw nothing, which makes me think the status about the truck connected is telling me porkies.
Going to go eat, been fighting with this for weeks, giving up for the night. Thanks for the help.
Paul
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Hi Paul,
We know the guys at VoIPOn well.
Try this guide for debugging captures:
http://www.surevoip.co.uk/support/wiki/using_command_line_tools_on_windows_and_gnu_linux#analysing_traffic (http://www.surevoip.co.uk/support/wiki/using_command_line_tools_on_windows_and_gnu_linux#analysing_traffic)
Is your firewall all setup OK?
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Hi Paul,
We know the guys at VoIPOn well.
Try this guide for debugging captures:
http://www.surevoip.co.uk/support/wiki/using_command_line_tools_on_windows_and_gnu_linux#analysing_traffic (http://www.surevoip.co.uk/support/wiki/using_command_line_tools_on_windows_and_gnu_linux#analysing_traffic)
Is your firewall all setup OK?
Thank you for the reply.
Yes my firewall is fine, and I have two other test IAX trunks working fine to external servers. (Sark boxes).
I had a brief look at your docs, will take a further look next week, have to go earn money so I can play later :-) Did get an email back from voipon. Seems it goes with the territory with voip and linux, everyone thinks we know as much as them :-)
We are sending the call to him, Below is the IAX packet as sent. We are not receiving any replies to the NEW packet:
Timestamp: 00014ms SCall: 00041 DCall: 00000 [My IP:4569]
VERSION : 2
CALLED NUMBER : 0845******
CODEC_PREFS : (alaw|ulaw|gsm|ilbc|g729|g723)
CALLING NUMBER :
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME :
LANGUAGE : en
USERNAME : 0845******
ENCRYPTION : 32768
FORMAT : 8
CAPABILITY : 40959
ADSICPE : 2
Now I note from the above that the USERNAME is not the username they gave me, but the actual 0845 number, I wonder? will try changing to the number and see what happens. I love the Sark system, but sometimes I get so confused as all the information on the net equates to standard asterisk conf files, and being a muppet I would not dare mess with them in a templated system.
Regards Paul
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How did you get on?
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How did you get on?
Well bless his little cotton socks, Mr Sark took a look at my system, and it seems that I have IAX blocked? which seems odd as I have had trunks running to other systems, so can only assume it was one of those open session out, then allow back?
Any hoo it is working, and thanks everyone for their help.
Regards Paul
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Hi All.
Have since updated to the latest ISO version SME8A14V31135.
Copied all the settings from the earlier working version of sail, but for the life of me I cannot get this trunk to register.
Connected to Asterisk 1.4.36 currently running on voipsbs (pid = 30707)
voipsbs*CLI> iax2 show registry
Host dnsmgr Username Perceived Refresh State
0.0.0.0:4569 N 747**** <Unregistered> 60 Unregistered
checked the obvious :-
[root@voipsbs ~]# /etc/init.d/masq status | grep 4569
ACCEPT udp -- 0.0.0.0/0 2**.***.***.*** udp dpt:4569
So I know IAX is open to the world.
All I can assume is that it is a miss configuration on my part, but I really am suffering with word blindness and cannot see it. In the trucks it shows as connected to the voip provider.
IAX2 show peers
peer8556/747*** 217.14.138.130 (s) 255.255.255.255 4569
Any suggestions?
Regards
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what does the registration string look like? - just type uuuu for the username and pppp for the password.
Kind Regards
S
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Hi
Under line in trunk the registration line is:-
747uuuu:pppppppp@iax.voipon.co.uk/74uuuu
And under peer :-
type=peer
host=iax.voipon.co.uk
qualify=3000
canreinvite=no
username=747uuuu
fromuser=747uuuu
secret=pppppppp
disallow=all
allow=alaw
allow=ulaw
Regards
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on the face of it, it looks OK. You'll need to ask voipon why it isn't accepting the registration.
Best
s
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It is something in my config. As I have just fired up the earlier version of Sail, and that registers fine?
Not a clue now on what to look for. Shall fire an email of to Voipon in the vain hope.
Thanks for the help.
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delete the trunk and redefine it. Make sure you get the userid and password on the first pass.
See if that cures it.
S
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Sadly same result.
I can make a test call to them fine, but for some reason it seems not to want to register with them on IAX for incoming. I have created this truck about 5 times, and religiously entered the username and password as was sent, even have gone so far as to cut and past from the working older system.
I am not a whiz with wireshark, so is there as way for me to see what is coming back from them?
As an aside, does this new version block outgoing and incoming SIP by default on the external interface (Server Gateway mode) or do you explicitly have to do an "config setprop sailSIP status disabled" ? Of course after doing the remoteaccess-update.
As even after that mine says :-
[root@voipsbs ~]# /etc/init.d/masq status | grep 5060
ACCEPT udp -- 0.0.0.0/0 (external IP) udp dpt:5060
Only reason I asked, as I noticed I was getting failed SIP request from a chancer, and that was after a short amount of time. Liked the idea of the OSSEC works well.
Regards
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to see what is running on the iax trunk; at the asterisk console you can do
iax2 set debug
turn it off with
iax2 set debug off
If your older system is still working OK it occurs to me that your router may be forwarding to the wrong IP address internally.
I tested the sailSIP enable/disable and it works just fine here. What does config show sailSIP give?
kind regards
s
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I did try the debug, but I cannot see anything coming from Voipon. So I changed the password for the working outgoing trunk, and then saw a registration error. However I see nothing coming in when from voipon if I make an incoming call via a normal pstn line to the voipon line.
I am in server gateway mode, this is a virtual server, running on the same server as my other virtual voip server (The old working one)
One thing perhaps I should mention, I have two other IAX trunks to friends Asterisk boxes that work fine.
As requested config show sailSIP & sailIAX
[root@voipsbs ~]# config show sailSIP
sailSIP=service
UDPPort=5060
access=public
status=enabled
[root@voipsbs ~]# config show sailIAX
sailIAX=service
UDPPort=4569
access=public
status=enabled
I wonder if I have the syntax right? config setprop sailSIP disabled? Will go check your site to make sure.
Regards
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If you aren't seeing anything in the debug then you will need to ask VoipOn why they aren't sending, and check your external firewall and port forwards to ensure they are correct.
config statement was correct in your previous post
config setprop sailSIP status disabled
Kind Regards
s
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Well I seem to be getting the run around with the trunk provider. It takes a while to get a reply and I am running out of time to get this working. What I could really do with is a monitor for incoming IAX packets, IAX debug does not show me any anything from the trunk provider unless I miss dial a number :-
[Apr 5 10:25:02] WARNING[3286]: chan_iax2.c:9197 socket_process: Call rejected by 217.14.138.130: No such context/extension
I can call their test line still. But still no incoming. Am I down to wireshark?
Regards
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RESOLVED
Well I am not sure why, but I found out what it was. When I created the truck I let SARK choose the names of the peer, I must have set the username of the "user" Now I set this to what my login was, ie username for Voipon. It turns out it wanted the incoming line number.
Hopefully might save someone some time.
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Glad you got it sorted Paul
Jeff