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<p class=MsoNormal><span style='color:#1F497D'>You can ignore this. I mistyped
the password, and once it was fixed, and registered correctly, both links
failed to work again.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>I have some extended information
from sip debug. Again, this shows up as soon as I try to register two connections.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><--- SIP read from
203.166.103.242:5060 ---><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>SIP/2.0 403 Forbidden<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Via: SIP/2.0/UDP
192.168.107.4:5060;branch=z9hG4bK454ad99d;received=59.167.248.154;rport=53487<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>From: "Joshua Small"
<sip:8001@192.168.107.4>;tag=as3d465ba3<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>To: <sip:phonnumber@gw02.mytel.net.au>;tag=as5937f41d<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Call-ID:
2f9f21865185cb9103ef86f438a79835@192.168.107.4<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>CSeq: 103 INVITE<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>User-Agent: Asterisk PBX<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Content-Length: 0<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div>
<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/">www.visinet.com.au</a></span><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:black'> </span><span
style='color:#1F497D'><o:p></o:p></span></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>
</div>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> Joshua Small <br>
<b>Sent:</b> Thursday, 13 September 2007 1:38 PM<br>
<b>To:</b> 'asterisk-users@lists.digium.com'<br>
<b>Subject:</b> FW: [asterisk-users] Problems with two trunks<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='color:#1F497D'>Update on this:<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>I found that by changing
insecure = very to insecure = invite, adding the second trunk no longer stopped
calls working.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>I’ve read the documentation on
this switch and still don’t see how it applies/is meant to get used.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Anyway, with this change in
place, the following may help:<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>asterisk*CLI> sip show
registry<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>Host
Username Refresh
State
Reg.Time<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>gw02.mytel.net.au:5060
11111
120 Request
Sent <o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>gw02.mytel.net.au:5060
22222
105 Registered Thu,
13 Sep 2007 23:33:47<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>I have set a dial plan so that
some handsets use the 2222 (not the real number) extension (which work)
and now I only need to determine why 11111 never seems to register.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>If I remove all traces of the
2222 connection from my config, 11111 registers fine.<o:p></o:p></span></p>
<div>
<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/">www.visinet.com.au</a></span><span
style='font-size:10.0pt;font-family:"Arial","sans-serif";color:black'> </span><span
style='color:#1F497D'><o:p></o:p></span></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>
</div>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Joshua
Small<br>
<b>Sent:</b> Thursday, 13 September 2007 10:44 AM<br>
<b>To:</b> asterisk-users@lists.digium.com<br>
<b>Subject:</b> [asterisk-users] Problems with two trunks<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<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/">www.visinet.com.au</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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