[asterisk-users] Problems with two trunks

[Paul Hales]

I would have usually used sip.conf or iax.conf - users.conf is not
something I know well....

PaulH


On Thu, 2007-09-13 at 10:44 +1000, Joshua Small wrote:
> Hi,
> 
>  
> 
> I am attempting to setup an asterisk server, current specs:
> 
> CentOS release 5 (Final)
> 
> Asterisk 1.4.11
> 
> Asterisk-gui checked out from SVN last week
> 
>  
> 
> I started with a fairly basic setup involving one VOIP provider who
> provided one dial in number, and a couple of handsets. Config files
> are below. It was pretty much totally built by Asterisk-gui, except
> for the fact I had to add “insecure=very” into users.conf in order to
> stop the dialin from our provider presenting an authentication error.
> Advice on any more correct approach would be appreciated, but is not
> the focus of this post:
> 
>  
> 
> Users.conf
> 
> ;several handsets setup like this...
> 
> [6001]
> 
> callwaiting = yes
> 
> context = numberplan-custom-1
> 
> email = jsmall at visinet.com.au
> 
> fullname = Joshua Small
> 
> hasagent = yes
> 
> hasdirectory = yes
> 
> hasiax = no
> 
> hasmanager = no
> 
> hassip = yes
> 
> hasvoicemail = no
> 
> host = dynamic
> 
> mailbox = 6001
> 
> secret = XXXXX
> 
> threewaycalling = yes
> 
> registeriax = no
> 
> registersip = yes
> 
> canreinvite = no
> 
> nat = no
> 
> dtmfmode = rfc2833
> 
> vmsecret = 1234
> 
>  
> 
> ;some PSTNS
> 
> [trunk_2]
> 
> callerid = asreceived
> 
> context = DID_trunk_2
> 
> group = 2
> 
> hasexten = no
> 
> hasiax = no
> 
> hassip = no
> 
> trunkname = Ports 1,2,3,4
> 
> trunkstyle = analog
> 
> zapchan = 1,2,3,4
> 
>  
> 
> ;my IP trunk
> 
> [trunk_3]
> 
> allow = all
> 
> context = DID_trunk_3
> 
> dialformat = ${EXTEN:1}
> 
> hasexten = no
> 
> hasiax = no
> 
> hassip = yes
> 
> host = gw02.mytel.net.au
> 
> port = 5060
> 
> registeriax = no
> 
> registersip = yes
> 
> secret = XXXXXXXX
> 
> trunkname = Custom - MyTel2
> 
> trunkstyle = customvoip
> 
> username = XXXXXXXX
> 
> type = friend
> 
> nat = yes
> 
>  
> 
> ;extensions.conf
> 
> [numberplan-custom-1]
> 
> plancomment = DialPlan1
> 
> include = default
> 
> include = parkedcalls
> 
> exten = _0XXXXX!,1,Macro(trunkdial,${trunk_3}/${EXTEN:1})
> 
> comment = _0XXXXX!,1,First,standard
> 
> ;a failover to PSTN, not yet enabled
> 
> ;exten = _0XXXXX!,2,Macro(trunkdial,${trunk_2}/0${EXTEN:1})
> 
> ;comment = _0XXXXX!,1,First,standard
> 
>  
> 
> At this point, everything appears to work fine. We can make calls from
> our several handsets using our voip link no problems.
> 
> We have two different accounts with our provider, the goal being
> certain handsets will connect to this account and therefore be billed
> separately. I haven’t gotten as far as to add the extra handsets and
> set an appropriate dialplan, all I did was add this to users.conf:
> 
>  
> 
> [trunk_extra]
> 
> allow = all
> 
> context = DID_trunk_3
> 
> dialformat = ${EXTEN:1}
> 
> hasexten = no
> 
> hasiax = no
> 
> hassip = yes
> 
> host = gw02.mytel.net.au
> 
> port = 5060
> 
> registeriax = no
> 
> registersip = yes
> 
> secret = XXXXXXXX
> 
> trunkname = Custom - MyTel Two
> 
> trunkstyle = customvoip
> 
> username = XXXXXXXXXX
> 
> type = friend
> 
> nat = yes
> 
>  
> 
> From this point on, my existing handsets don’t appear to be able to
> get a line out. My console looks like this (from the first call out):
> 
> Connected to Asterisk 1.4.11 currently running on asterisk (pid =
> 31999)
> 
>     -- Remote UNIX connection
> 
> Verbosity is at least 8
> 
>     -- Executing [00425298582 at numberplan-custom-1:1]
> Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new
> stack
> 
>     -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20",
> "SIP/trunk_3/0425298582") in new stack
> 
>     -- Called trunk_3/0425298582
> 
> [Sep 13 20:42:16] WARNING[32011]: chan_sip.c:12016
> handle_response_invite: Received response: "Forbidden" from '"Joshua
> Small" <sip:8001 at 192.168.107.4>;tag=as29bb274d'
> 
>     -- SIP/trunk_3-097ac708 is circuit-busy
> 
>   == Everyone is busy/congested at this time (1:0/1/0)
> 
>     -- Executing [s at macro-trunkdial:2] Goto("SIP/8001-b7d0bb20",
> "s-CONGESTION|1") in new stack
> 
>     -- Goto (macro-trunkdial,s-CONGESTION,1)
> 
>     -- Executing [s-CONGESTION at macro-trunkdial:1]
> NoOp("SIP/8001-b7d0bb20", "") in new stack
> 
>   == Auto fallthrough, channel 'SIP/8001-b7d0bb20' status is
> 'CONGESTION'
> 
>  
> 
>  
> 
> Any advice on why our trunk_3 becomes congested, just because
> trunk_extra is set to exist, is appreciated.
> 
> Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887
> 959 | www.visinet.com.au 
> 
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> 
>  
> 
> 
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