Setting up trunks - can't find where to config?

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Setting up trunks - can't find where to config?

Postby olsjohnluke » Fri Jan 05, 2007 1:45 pm

Have a TDM400 with 2 FXO and 2 FXS ports each (the FXO's are ch3/4). For a while, AsteriskNow was cfg'd to route all outgoing calls through the FXS lines, so every time I'd call from a VOIP phone, an analogue handset would ring. I didn't understand how, but I didn't want it to, so I disabled channels 1 and 2, and am trying to configure things so that outgoing calls get routed out the PSTN lines. Now, every time I make a call, I get:

Code: Select all
    -- Executing [9930600@numberplan-custom-1:1] Macro("SIP/202-08201af0", "trunkdial|/9930600") in new stack
    -- Executing [s@macro-trunkdial:1] Dial("SIP/202-08201af0", "/9930600") in new stack
[Jan  5 11:28:38] WARNING[2596]: channel.c:2910 ast_request: No channel type registered for ''
[Jan  5 11:28:38] WARNING[2596]: app_dial.c:1081 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-trunkdial:2] Goto("SIP/202-08201af0", "s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-trunkdial,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-trunkdial:1] NoOp("SIP/202-08201af0", "") in new stack
[Jan  5 11:28:48] WARNING[2596]: pbx.c:2460 __ast_pbx_run: Timeout, but no rule 't' in context 'numberplan-custom-1'
asterisk*CLI>


From what I can interpret in that, it's officially got no outgoing trunks configured, so it doesn't know where to go (leading to CHANUNAVAIL). I've looked over extensions.conf, zapata.conf, users.conf, etc, and everything seems to be right, except it's obviously not because no calls are being routed properly. I'm at my wits end here, and every tutorial I read in attempts to decipher what I'm looking at seem to complicate things further.

I have this section in my users.conf file:
Code: Select all
[trunk_1]
; secret =
; provider = iaxtel
zapchan = 3,4
trunkstyle = analog
; username =
trunkname = Ports 3,4
callerid = "Companyname" <3609930600>
hasexten = no
hassip = no
hasiax = no
; registeriax =
; registersip =
host = dynamic
dialformat =
context =
group = undefined
insecure =
fromuser =


I commented out the IAX and SIP lines because they're PSTN - did this make a difference? Should I not have? Since the values were blank anyway, I don't think it would have impacted it (also I know that the CID values won't get updated because we're not using a PRI yet - this is a demo run before we bring over a couple of T1s). And also, I can't figure out where trunk_1 gets configured... what config file and section (with AsteriskNow) sets Trunk_1 to equal FXO ports 3/4?

I'm faced with the maddening challenge of broadly understanding where problem lies, but not having the first idea how to fix it. I would greatly appreciate any help anyone can offer!

Rgds,
John
olsjohnluke
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Postby bkruse » Fri Jan 05, 2007 1:53 pm

John, the GUI writes to users.conf

You CAN write to it, however, I recommend taking a look at this link:

http://www.voip-info.org/wiki/index.php ... g+sip.conf

Just for now, as the GUI is not 100% with setting up providers, but it will get better, I promise. :]

But for now, just editing your sip.conf, and register => to your sip provider.


Hope this helps!

Hopefully, in the near future, you will be able to set it all up from the gui :]

-bkruse
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Postby olsjohnluke » Fri Jan 05, 2007 2:00 pm

Appreciate the quick reply, but I'm a little confused. I'm not using a sip provider - it's VOIP internally (Polycom SP501 and Xlite phones), and 2 PSTN lines externally. Is sip.conf where I need to be working out of anyway?

Thanks -
olsjohnluke
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Posts: 28
Joined: Fri Dec 01, 2006 3:36 pm

Postby bkruse » Fri Jan 05, 2007 2:15 pm

olsjohnluke, I am sorry for not reading your post all the way through, sorry :]

yes, sip.conf is where its at, IF you cannot get it working in the GUI.

check out that link i sent you, and first, start by pointing your polycom at your asterisk server, and set username and password to woot and blah.

then, in your asterisk server, navigate to /etc/asterisk and (fav text editor here) sip.conf

add these lines:
Code: Select all
[woot]
username=woot
secret=blah
context=default ; (put a context you want the phone to go to when you pick it up, has to be one defined in extensions.conf or nothing will work ahh!)
type=friend
;disallow=all
allow=all ; (codecs here, check on your polycom phone, if you dont know, remove the disallow=all line)
; nat settings
qualify=10000 ; (set if your behind a nat)
nat=yes ; (only if your phone cannot directly reach your * box)


Hope this helps!
-bkruse
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Postby olsjohnluke » Fri Jan 05, 2007 2:30 pm

I think we're getting closer, but I'm still a little confused....

At http://forums.digium.com/viewtopic.php? ... ght=#39934 it was pointed out to me that the GUI writes/reads only from users.conf. All my extension info is plugged into users.conf, that happened automatically. I can call internally from extension to extension without any issue - the phones authenticate and register just fine. It's the routing to an outgoing trunk that doesn't seem to be happening, because the outgoing trunk doesn't seem to be configured, despite the fact it appears in zapata.conf, and at one point it was routing (incorrectly) to zap 1-1. Is that a context issue? Can the context issue be resolved by modifying users.conf?

In short - how do I set my extensions to call out thru the FXO lines?

Befuddled,
John
olsjohnluke
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Posts: 28
Joined: Fri Dec 01, 2006 3:36 pm

Postby bkruse » Fri Jan 05, 2007 2:42 pm

ok, thats fine.

I suggest from here(and sorry it writes only provider/phone information to users.conf, not iax.conf and sip.conf is what i meant)

so....if you are registering well, try editing extensions.conf to dial a zap line
or a zap group at that
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Postby j4m3s » Fri Jan 05, 2007 6:45 pm

Can you please log this bug into bugs.digium.com under the GUI? Thanks!
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Postby kimbroy » Tue Jan 09, 2007 12:13 am

This is the same problem I am having with my outgoing calls...

You would think Digium would make this beta work with their cards ...

My friend got it working with his itsp..oh boy what fun..


Oh well hope there is a fix...
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Postby olsjohnluke » Wed Jan 10, 2007 1:57 pm

j4m3s wrote:Can you please log this bug into bugs.digium.com under the GUI? Thanks!


Willdo. Thx.
olsjohnluke
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Posts: 28
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