Support required in configuring SIP.conf


#1

Hello Everyone,
I have installed asterisk on my slackware 10.2, i installed all the prerequites and thereafter asterisk through CVS, and it has been installed successfully and seems to be working.
Now my requirement is as follows:
I have a Analog VOIP Gateway(works on SIP), and this server on which i have installed asterisk, now i want to know what configuration is required to be done so that i can accept a call from this analog gateway and also what configuration is required to be done to further dial a number via VOIP itself. i do not have any digium card installed on my machine.
All the replies would be appreciated.

Regards

Vijay Gandhi


#2

Vijay,

There are alot of people on this board that can and will help you if you run into a problem but there is an unspocken expectation that you will take it upon yourself to atleast try to learn some of the basics. Read the wiki and the book (links below). Configure your extensions.conf and sip.conf and your gateway. Post back here if you run into any snags along the way.

-Eric


#3

Hi Guys,
I have configured SIP.conf as follows:

Context = from-agent
port=5060
bindaddr=0.0.0.0
srvlookup=yes

rtptimeout =1200
rtpholdtimeout=1200
dtmfmode=inband
disllow=all
allow=g729

user=CISCO
[101]
type=friend
username=101
secret=101
host=dynamic
defaultip=192.168.0.1 # IP for my VOIP Gateway.
dtmfmode=inband
canreinvite=no
context=from-agent

and so on for all the extentions till 124 (from 101 to 124)

and for dialing out the number

[ECFC]
type=friend
secret=
host=203.122.28.108
dtmfmode=rfc2833
fromdomain=203.122.28.108
fromuser=NODID
nat=no
canreinvite=no
context=from-agent.

this is my configuration for SIP.conf

and in extensions.conf, here is the conf.

[general]
static=yes
writeprotect=no
[globals]
[from-manager]
EXTEN=>9000,1,Conference(conf|M|1)
EXTEN=>1000,1,AMD(1.5|and-${UNIQUEID}-${TIMESTAMP})
EXTEN=>1000,2,Queue(${QUEUE_NAME})
EXTEN=>1000,3,Dial(${INTERFACE})
EXTEN=>1000,4,goto(2)
EXTEN=>2000,1,Conference(${Conference_Name}/$Conference_mode|1)
EXTEN=>4000,1,PlayBack(${Plyabck_file})
EXTEN=>5000,1,Playback(moh)
EXTEN=>5000,2,goto(1)
EXTEN=>6000,1,Dial(${CUSTOMER_CHANNEL})
EXTEN=>7000,1,Dial(${IVR})
EXTEN=>7000,2,wit(999)

[from-agent]
include=>outgoing
include=>internal
EXTEN=>h,1,Hangup()
EXTEN=>t,1,Hangup()

[internal]
EXTEN=>-.,1,Dial(SIP/${EXTEN})
[outgoing]
EXTEN=>-9.,1,Dial(SIP/64#1${EXTEN:2}@ECFC)

and ZAPATA.conf has been configured as follows:

[trunkgroups]
[channel]
context=from-agent
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=no
callwaitingcallerid=no
threewaycalling=no
transfer=no
cancellforward=no
callreturn=no
echocancel=no
echocancelwhenbridged=no
relaxdtmf=yes
rxgain=2.0
txgain=2.0
immediate=no
signalling=pri_cpe
switchtype=euroisdn
group=1
channel=>1-15,17-31,32-46,48-62
Signalling=pri_net
pritimer=N200,10
pritimer=T305,300
pritimer=T308,2000
switchtype=euroisdn
group=2
channel=>63-77,79-93,94-108,110-124

Using these configurations i am able to compile &run ASTERISK properly, it shown me cli>, but when i pick up the phone attached to my gateway and dial any number, it does not shows anything, no logs and nothing, please update what can be wrong in the conf.

Regards


#4

Add more verbosity to asterisk, then make the call. IE: asterisk -vvvvvvr (if it is already running)