[asterisk-users] FW: Problems with two trunks
Paul Hales
pdhales at optusnet.com.au
Thu Sep 13 01:03:35 CDT 2007
Does the mytel gateway show up fine in sip show peers?
PaulH
On Thu, 2007-09-13 at 15:06 +1000, Joshua Small wrote:
> You can ignore this. I mistyped the password, and once it was fixed,
> and registered correctly, both links failed to work again.
>
> I have some extended information from sip debug. Again, this shows up
> as soon as I try to register two connections.
>
>
>
> <--- SIP read from 203.166.103.242:5060 --->
>
> SIP/2.0 403 Forbidden
>
> Via: SIP/2.0/UDP
> 192.168.107.4:5060;branch=z9hG4bK454ad99d;received=59.167.248.154;rport=53487
>
> From: "Joshua Small" <sip:8001 at 192.168.107.4>;tag=as3d465ba3
>
> To: <sip:phonnumber at gw02.mytel.net.au>;tag=as5937f41d
>
> Call-ID: 2f9f21865185cb9103ef86f438a79835 at 192.168.107.4
>
> CSeq: 103 INVITE
>
> User-Agent: Asterisk PBX
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>
> Content-Length: 0
>
>
>
> Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887
> 959 | www.visinet.com.au
>
> 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.
>
>
>
>
> From: Joshua Small
> Sent: Thursday, 13 September 2007 1:38 PM
> To: 'asterisk-users at lists.digium.com'
> Subject: FW: [asterisk-users] Problems with two trunks
>
>
>
>
> Update on this:
>
>
>
> I found that by changing insecure = very to insecure = invite, adding
> the second trunk no longer stopped calls working.
>
> I’ve read the documentation on this switch and still don’t see how it
> applies/is meant to get used.
>
>
>
> Anyway, with this change in place, the following may help:
>
>
>
> asterisk*CLI> sip show registry
>
> Host Username Refresh State
> Reg.Time
>
> gw02.mytel.net.au:5060 11111 120 Request
> Sent
>
> gw02.mytel.net.au:5060 22222 105 Registered
> Thu, 13 Sep 2007 23:33:47
>
>
>
> 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.
>
>
>
> If I remove all traces of the 2222 connection from my config, 11111
> registers fine.
>
> Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887
> 959 | www.visinet.com.au
>
> 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.
>
>
>
>
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua
> Small
> Sent: Thursday, 13 September 2007 10:44 AM
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] Problems with two trunks
>
>
>
>
> 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
>
> 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.
>
>
>
>
> _______________________________________________
>
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