Duane,
Thanks for the input guys.
I give up on SIP already. IAX trunking registers but it seems like a dialplan
issue now.
Here is what I am getting from ServerA (when trying to reach ServerB):
-- Executing [...@default:1] Dial("SIP/101-08b21558", "IAX2/trunk_demo/601") in
new stack
-- Called trunk_demo/601
[May 23 21:33:20] WARNING[17777]: chan_iax2.c:7984 socket_process: Call
rejected by 206.206.206.206: No such context/extension
-- Hungup 'IAX2/trunk_demo-16384'
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/101-08b21558' status is 'CHANUNAVAIL'
ServerB has no verbose output. It does have debug output though.
ServerA is running Asterisk 1.4 and serverB is running Elastix (Asterisk 1.4 as
well). There seems to be something odd with Elastix that is not taking in
calls. Here is the trunk peer info:
*******************************************************************
Peer Details:
host=122.122.122.122
username=trunk_demo
secret=trunk_d3m0
type=friend
context=from-trunk
User Context: user
User Details:
host=122.122.122.122
type=peer
context=from-trunk
Register String: trunk_demo:[email protected]
***********************************************************************
ServerA has the following trunk info:
[general]
port=4569
bandwidth=low
disallow=all
allow=gsm
nat=yes
jitterbuffer=yes
tos=lowdelay
;context=from-trunk
register => trunk_demo:[email protected]
[trunk_demo]
type=friend
host=dynamic
trunk=yes
auth=plaintext
secret=trunk_d3m0
username=trunk_demo
qualify=yes
nat=yes
context=from-trunk
Dialplan on ServerA has the following which gets the CLI noted at the begining
of the e-mail:
[general]
static=yes
writeprotect=no
[macro-AnsMachine]
exten => s,1,AMD()
[inbound]
exten => _72.,1,Ringing()
exten => _72.,2,Wait(2)
exten => _72.,3,Hangup()
exten => h,1,Set(CALLERID(num)}=${CALLERID(num)})
exten => h,2,Dial(SIP/7...@asterisk,,tTo)
[default]
exten => _99.,1,Playback(demo-instruct)
exten => _88.,1,Dial(SIP/6...@asterisk)
exten => 55,1,Dial(IAX2/trunk_demo/601)
exten => s,1,Playback(welcome)
Thanks,
Bruce
P.S. Note that the IPs are changed. Don't try them :)
> Date: Sat, 23 May 2009 23:39:53 +1000
> From: [email protected]
> To: [email protected]
> Subject: Re: [on-asterisk] VoIP banned?
>
> Bruce N wrote:
>> Hi Guys,
>> I have a simple SIP extension setup on a server which I can register too
>> just fine using eyeBeam. Someone in India is trying to register and they get
>> timeout and here is what I see from CLI:
>> -- Registered SIP '4044567260' at 122.122.122.122 port 27282 expires 3600 --
>> Saved useragent "X-Lite release 1011s stamp 41150" for peer 4044 --
>> Unregistered SIP '4044' -- Unregistered SIP '4044' -- Unregistered SIP '4044'
>>
>> And it goes on, keeps registering and unregistering. It's never able to make
>> calls and it never shows registered on xlite. Anyone seen this behavior? and
>> why is the port showing 27282? shouldn't that be 5060?
>> Could this be due to VoIP being banned? or due to the router that is
>> supplied by their provider?
>
> Sounds more like a UDP timeout issue on your NAT router, and the client
> can use any port only the server usually needs to use 5060.
>
> Some NAT implementations just plain break UDP connections because they
> are connectionless etc.
>
> You might be able to get away with IAX, but it uses UDP as well so that
> may still not work.
>
> Problems like this can usually be gotten round by doing a port forward
> on the NAT router and setting a fixed port in the client software if
> possible.
>
> --
>
> Best regards,
> Duane
>
> http://www.freeauth.org - Enterprise Two Factor Authentication
> http://www.nodedb.com - Think globally, network locally
> http://www.sydneywireless.com - Telecommunications Freedom
> http://e164.org - Global Communication for the 21st Century
>
> "In the long run the pessimist may be proved right,
> but the optimist has a better time on the trip."
>
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