Unable to make Outbound Calls

I have Asterisk 1.2.10 up and running on Cent OS with a Digium TE110P T1 card. I am able to make calls from extension to extension within the company. I am also able to make inbound calls from the outside to the extensions.

Currently I have 1 Aastra 9133i working and registering just fine. Voicemail works as well.

I need help in configuring to make outbound calls. I don’t need any special configs, just the ability to make an outgoing call. Here is a peek at my configs:

zapata.conf
[channels]
group=1
switchtype=national
context=incoming
signalling=pri_cpe
channel => 1-23
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
callerid=asreceived

extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[incoming]
exten => _3699,1,Dial(SIP/3699,20)
exten => _3699,2,VoiceMail(3699@default)
exten => _3699,3,PlayBack(vm-goodbye)
exten => _3699,4,Wait(2)
exten => _3699,5,HangUp()
;
exten => 9999,1,VoiceMailMain

sip.conf
[general]
context=incoming
srvlookup=yes
bindport=5060
bindaddr=0.0.0.0
[3699]
type=friend
username=3699
secret=123456
qualify=yes
nat=no
host=dynamic
canreinvite=no
mailbox=3699@voicemail

Any help would be greatly appreciated. Once I get the outbound figured out then I can fine tune it from there.

Thanks all

…Razr

a few pointers …

  1. read the sticky at the top of the forum, especially the one about the book Asterisk - TFOT … particularly useful when starting out.

  2. your SIP users will need a context definition if you want to maintain separation between incoming and outgoing callers.

  3. your zapata.conf … the channel= line should be the last line … it inherits everything above it.

so, add a context=outgoing to your SIP users, create an [outgoing] context to extensions.conf, then a line such as exten=> _XXXXX.,1,Dial(ZAP/g1/${EXTEN},,TW) would get you started/calling.

look in the wiki for a more complete example.

I did as you suggested and this is what shows up in the CLI when I attempt to make an outbound call:

== Parsing ‘/etc/asterisk/codecs.conf’: Found
– codec_g726: using generic PLC
– Executing Dial(“SIP/3699-0919e280”, “ZAP/g1/6912130||TW”) in new stack
– Requested transfer capability: 0x00 - SPEECH
– Called g1/6912130
– Channel 0/1, span 1 got hangup
– Hungup ‘Zap/1-1’
== Everyone is busy/congested at this time (1:0/0/1)
– Executing Hangup(“SIP/3699-0919e280”, “”) in new stack
== Spawn extension (outgoing, 6912130, 2) exited non-zero on 'SIP/3699-0919e280’
Asterisk*CLI>

I have the book in front of me, but I get lost with the variables, I think reading it several times has actually confused me more then helped me.

Could this actually be a Telco issue? The line shows as busy/congested…

Thanks

…Razr

bump