vicidial.org

Pu the carrier as normal in admin / carriers and test it.

[CONTEXTHERE]
host=69.30.XX.XXX
context=trunkinbound
type=friend
qualify=1000
dtmfmode=rfc2833
canreinvite=no
allow=ulaw
allow=gsm
allow=g729
port=5060

obviously only "allow" codecs you actually want

note: no "user" or "username" or other auth entries at all.

hi people ,
thanks for ur support in here ...
he says its a termination account ..and has sent me this config
[work]
type=peer
dtmfmode=rfc2833
nat=no
promiscredir=yes
canreinvite=no
host=69.30.55.34
disallow=all
allow=alaw
allow=gsm
allow=ulaw
and the dial plan entry is

[403]
exten => 403,1,Answer()
exten => 403,2,wait(3)
exten => 403,3,Playback(tt-monkeys)
exten => 403,4,dial(sip/403)
exten => 403,5,Hangup

exten => _011X.,1,NoOP()
exten => _011X.,n,answer()
exten => _011X.,n,set(CALLERID(all)=15037109111)
exten => _011X.,n,sipdtmfmode(rfc2833)
exten => _011X.,n,dial(SIP/work/${EXTEN})

exten => _1X.,1,NoOP()
exten => _1X.,n,answer()
exten => _1X.,n,set(CALLERID(all)=12094655000)
exten => _1X.,n,sipdtmfmode(rfc2833)
exten => _1X.,n,dial(SIP/work/${EXTEN})

i dont understand .. please help
i just want to make calls

CarrierID: sipcarrier1

Carrier Name: sipcarrier1

Registration String: BLANK

Template ID: --NONE--

Account Entry:
[sipcarrier1]
type=peer
dtmfmode=rfc2833
nat=no
promiscredir=yes
canreinvite=no
host=69.30.55.34
disallow=all
allow=alaw
allow=gsm
allow=ulaw

Protocol: SIP

Globals String: DIAL9SERVICE = SIP/sipcarrier1

Dialplan Entry:
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${DIAL9SERVICE}/${EXTEN:1},,tTor)
exten => _91NXXNXXXXXX,3,Hangup

Server IP: (your server)
Active: Y

In the Campaign: Dial Prefix: 9
Campaign CallerID: 15037109111

hi ,,
account entry is
[sipcarrier1]
type=peer
dtmfmode=rfc2833
nat=no
promiscredir=yes
canreinvite=no
host=69.30.55.34
disallow=all
allow=alaw
allow=gsm
allow=ulaw

dial plan entry is
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${DIAL9SERVICE}/${EXTEN:1},,tTor)
exten => _91NXXNXXXXXX,3,Hangup


and the error i am getting is when i place a calll is
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----2
1-----CONGESTION---------- completed, returning 0
-- Executing AGI("SIP/cc101-b7a2c420", "agi://127.0.0.1:4577/call_log") in n
ew stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/cc101-b7a2c420", "SIP/sipcarrier1/15082212561||tTor")
in new stack
-- Called sipcarrier1/15082212561
Jun 18 18:39:35 WARNING[6630]: chan_sip.c:9890 handle_response_invite: Forbidden
- wrong password on authentication for INVITE to '"cc101" <sip:cc101@122.169.25
3.125>;tag=as2433bb77'
-- SIP/sipcarrier1-09909740 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("SIP/cc101-b7a2c420", "") in new stack
== Spawn extension (default, 915082212561, 3) exited non-zero on 'SIP/cc101-b7
a2c420'
-- Executing DeadAGI("SIP/cc101-b7a2c420", "agi://127.0.0.1:4577/call_log--H
Vcauses--PRI-----NODEBUG-----21-----CONGESTION----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----2
1-----CONGESTION---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
what to do now ...

if you are behind a firewall you likely need to set your "externip=" in "sip.conf"

hi there ,
thank u for ur quick response...
can u elaborate what ip i shoul give in extern ip i mean should i give my server ip or the ip of the carrier ..in here
thank u in advance

externip is for the ip address on the outside of your firewall.

your asterisk server thinks it has an ip, and it broadcasts that ip in the sip packets. unfortunately, THAT ip is a "local" ip address, and it is not accessible by the provider.

the provider uses this as a security protocol, the ip inside the sip packet IS your password. so you must change that to the ip address they have on file for your server (which is your EXTERNAL IP or "externip").

so putting your externip as your external IP address puts the external ip address inside your sip packets and essentially allows each packet to "authenticate" as it travels. removing the need for a user/password in the sip context.