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<p class=MsoNormal>Hi,<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>I am attempting to setup an asterisk server, current specs:<o:p></o:p></p>
<p class=MsoNormal>CentOS release 5 (Final)<o:p></o:p></p>
<p class=MsoNormal>Asterisk 1.4.11<o:p></o:p></p>
<p class=MsoNormal>Asterisk-gui checked out from SVN last week<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>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:<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Users.conf<o:p></o:p></p>
<p class=MsoNormal>;several handsets setup like this...<o:p></o:p></p>
<p class=MsoNormal>[6001]<o:p></o:p></p>
<p class=MsoNormal>callwaiting = yes<o:p></o:p></p>
<p class=MsoNormal>context = numberplan-custom-1<o:p></o:p></p>
<p class=MsoNormal>email = jsmall@visinet.com.au<o:p></o:p></p>
<p class=MsoNormal>fullname = Joshua Small<o:p></o:p></p>
<p class=MsoNormal>hasagent = yes<o:p></o:p></p>
<p class=MsoNormal>hasdirectory = yes<o:p></o:p></p>
<p class=MsoNormal>hasiax = no<o:p></o:p></p>
<p class=MsoNormal>hasmanager = no<o:p></o:p></p>
<p class=MsoNormal>hassip = yes<o:p></o:p></p>
<p class=MsoNormal>hasvoicemail = no<o:p></o:p></p>
<p class=MsoNormal>host = dynamic<o:p></o:p></p>
<p class=MsoNormal>mailbox = 6001<o:p></o:p></p>
<p class=MsoNormal>secret = XXXXX<o:p></o:p></p>
<p class=MsoNormal>threewaycalling = yes<o:p></o:p></p>
<p class=MsoNormal>registeriax = no<o:p></o:p></p>
<p class=MsoNormal>registersip = yes<o:p></o:p></p>
<p class=MsoNormal>canreinvite = no<o:p></o:p></p>
<p class=MsoNormal>nat = no<o:p></o:p></p>
<p class=MsoNormal>dtmfmode = rfc2833<o:p></o:p></p>
<p class=MsoNormal>vmsecret = 1234<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>;some PSTNS<o:p></o:p></p>
<p class=MsoNormal>[trunk_2]<o:p></o:p></p>
<p class=MsoNormal>callerid = asreceived<o:p></o:p></p>
<p class=MsoNormal>context = DID_trunk_2<o:p></o:p></p>
<p class=MsoNormal>group = 2<o:p></o:p></p>
<p class=MsoNormal>hasexten = no<o:p></o:p></p>
<p class=MsoNormal>hasiax = no<o:p></o:p></p>
<p class=MsoNormal>hassip = no<o:p></o:p></p>
<p class=MsoNormal>trunkname = Ports 1,2,3,4<o:p></o:p></p>
<p class=MsoNormal>trunkstyle = analog<o:p></o:p></p>
<p class=MsoNormal>zapchan = 1,2,3,4<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>;my IP trunk<o:p></o:p></p>
<p class=MsoNormal>[trunk_3]<o:p></o:p></p>
<p class=MsoNormal>allow = all<o:p></o:p></p>
<p class=MsoNormal>context = DID_trunk_3<o:p></o:p></p>
<p class=MsoNormal>dialformat = ${EXTEN:1}<o:p></o:p></p>
<p class=MsoNormal>hasexten = no<o:p></o:p></p>
<p class=MsoNormal>hasiax = no<o:p></o:p></p>
<p class=MsoNormal>hassip = yes<o:p></o:p></p>
<p class=MsoNormal>host = gw02.mytel.net.au<o:p></o:p></p>
<p class=MsoNormal>port = 5060<o:p></o:p></p>
<p class=MsoNormal>registeriax = no<o:p></o:p></p>
<p class=MsoNormal>registersip = yes<o:p></o:p></p>
<p class=MsoNormal>secret = XXXXXXXX<o:p></o:p></p>
<p class=MsoNormal>trunkname = Custom - MyTel2<o:p></o:p></p>
<p class=MsoNormal>trunkstyle = customvoip<o:p></o:p></p>
<p class=MsoNormal>username = XXXXXXXX<o:p></o:p></p>
<p class=MsoNormal>type = friend<o:p></o:p></p>
<p class=MsoNormal>nat = yes<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>;extensions.conf<o:p></o:p></p>
<p class=MsoNormal>[numberplan-custom-1]<o:p></o:p></p>
<p class=MsoNormal>plancomment = DialPlan1<o:p></o:p></p>
<p class=MsoNormal>include = default<o:p></o:p></p>
<p class=MsoNormal>include = parkedcalls<o:p></o:p></p>
<p class=MsoNormal>exten = _0XXXXX!,1,Macro(trunkdial,${trunk_3}/${EXTEN:1})<o:p></o:p></p>
<p class=MsoNormal>comment = _0XXXXX!,1,First,standard<o:p></o:p></p>
<p class=MsoNormal>;a failover to PSTN, not yet enabled<o:p></o:p></p>
<p class=MsoNormal>;exten = _0XXXXX!,2,Macro(trunkdial,${trunk_2}/0${EXTEN:1})<o:p></o:p></p>
<p class=MsoNormal>;comment = _0XXXXX!,1,First,standard<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>At this point, everything appears to work fine. We can make
calls from our several handsets using our voip link no problems.<o:p></o:p></p>
<p class=MsoNormal>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:<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>[trunk_extra]<o:p></o:p></p>
<p class=MsoNormal>allow = all<o:p></o:p></p>
<p class=MsoNormal>context = DID_trunk_3<o:p></o:p></p>
<p class=MsoNormal>dialformat = ${EXTEN:1}<o:p></o:p></p>
<p class=MsoNormal>hasexten = no<o:p></o:p></p>
<p class=MsoNormal>hasiax = no<o:p></o:p></p>
<p class=MsoNormal>hassip = yes<o:p></o:p></p>
<p class=MsoNormal>host = gw02.mytel.net.au<o:p></o:p></p>
<p class=MsoNormal>port = 5060<o:p></o:p></p>
<p class=MsoNormal>registeriax = no<o:p></o:p></p>
<p class=MsoNormal>registersip = yes<o:p></o:p></p>
<p class=MsoNormal>secret = XXXXXXXX<o:p></o:p></p>
<p class=MsoNormal>trunkname = Custom - MyTel Two<o:p></o:p></p>
<p class=MsoNormal>trunkstyle = customvoip<o:p></o:p></p>
<p class=MsoNormal>username = XXXXXXXXXX<o:p></o:p></p>
<p class=MsoNormal>type = friend<o:p></o:p></p>
<p class=MsoNormal>nat = yes<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>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):<o:p></o:p></p>
<p class=MsoNormal>Connected to Asterisk 1.4.11 currently running on asterisk
(pid = 31999)<o:p></o:p></p>
<p class=MsoNormal> -- Remote UNIX connection<o:p></o:p></p>
<p class=MsoNormal>Verbosity is at least 8<o:p></o:p></p>
<p class=MsoNormal> -- Executing
[00425298582@numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20",
"trunkdial|SIP/trunk_3/0425298582") in new stack<o:p></o:p></p>
<p class=MsoNormal> -- Executing [s@macro-trunkdial:1]
Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582") in new
stack<o:p></o:p></p>
<p class=MsoNormal> -- Called trunk_3/0425298582<o:p></o:p></p>
<p class=MsoNormal>[Sep 13 20:42:16] WARNING[32011]: chan_sip.c:12016
handle_response_invite: Received response: "Forbidden" from
'"Joshua Small" <sip:8001@192.168.107.4>;tag=as29bb274d'<o:p></o:p></p>
<p class=MsoNormal> -- SIP/trunk_3-097ac708 is circuit-busy<o:p></o:p></p>
<p class=MsoNormal> == Everyone is busy/congested at this time (1:0/1/0)<o:p></o:p></p>
<p class=MsoNormal> -- Executing [s@macro-trunkdial:2]
Goto("SIP/8001-b7d0bb20", "s-CONGESTION|1") in new stack<o:p></o:p></p>
<p class=MsoNormal> -- Goto (macro-trunkdial,s-CONGESTION,1)<o:p></o:p></p>
<p class=MsoNormal> -- Executing
[s-CONGESTION@macro-trunkdial:1] NoOp("SIP/8001-b7d0bb20",
"") in new stack<o:p></o:p></p>
<p class=MsoNormal> == Auto fallthrough, channel 'SIP/8001-b7d0bb20'
status is 'CONGESTION'<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Any advice on why our trunk_3 becomes congested, just
because trunk_extra is set to exist, is appreciated.<o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:red'>Joshua
Small</span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:black'> | Senior Network Engineer | <b>VisiNet</b> | P. +61
1300 887 959 | </span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'><a href="http://www.visinet.com.au/"><span style='color:blue'>www.visinet.com.au</span></a></span><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:black'> </span><o:p></o:p></p>
<p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:#666666'>This
e-mail is intended for use by the named recipients only and contains
confidential information. Opinions and other information in this message that
pertain to the sender's employer and its products and services represent the
opinion of the sender and not necessarily those of the employer. </span><span
style='font-size:10.0pt;font-family:"Times New Roman","serif";color:#666666'><o:p></o:p></span></p>
<p class=MsoNormal><o:p> </o:p></p>
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