vicidial.org

Hi Everyone,

I have installed VicidialNow and configured sip.conf and extensions.conf accordingly.

My Asterisk server is behind NAT and sip clients are also in same network which is behind NAT.

Port 5060, is blocked in my country, so I used 5744 for Registration with my SIP provider, as they do support 5744 port. When I done that I was able to register my Asterisk box with my SIP Provider. I can say that because when I run command "sip show registry", it shows that its registered with SIP provider.

I have created extensions and configured them on Eyebeam softphone and they also registered successfully with Asterisk box. I even try to make calls between extensions and it goes fine.

Now I am trying to make an outbound call (Manual) through eyebeam softphone, it seems that its doing something but after short period I am getting Error message on EyeBeam saying "603 Declined".

Here is my sip.conf file:-

[general]
bindport = 5744
bindaddr = 0.0.0.0
context = default

; register SIP account on remote machine if using SIP trunks
register => my_username:my_password@VoIPProvider's_IP_Add:5744
; setup account for SIP trunking:

[SIPtrunk]
type=friend
host=VoIPProvider's_IP_Add
fromdomain=VoIPProvider's_IP_Add
fromuser=almtech
secret=my_password
username=my_username
context=default
authname=my_username
;dtmfmode=inband
canreinvite=yes
qualify=no
nat=yess

[801]
disallow=all
allow=ulaw
allow=alaw
allow=g729
type=friend
username=801
secret=1234
host=dynamic
dtmfmode=inband
qualify=no
nat=yes

; same settings for the rest of the extensions

Here is my extensions.conf:-
[general]
static=yes
writeprotect=no

[globals]
SIPtrunk=SIP/my_username:my_password@VoIPProvider's_IPADD

[default]

----many other extensions which I make them comments......................but the below one I have added for calling USA----

; Example phone extensions

; dial a long distance outbound number through a SIP provider
exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _1NXXNXXXXXX,2,Dial(sip/${EXTEN}@SIPtrunk,30,o)
exten => _1NXXNXXXXXX,3,Hangup

Now I am stuck what to do. I have tried several things but no use.

Can anyone help me with this problem "603 Declined". I know its not a Vicidial problem but I am unable to get answer from other forums that is why posting it in this forum.

Thanks & Regards

ATIF

Here is my Asterisk CLI output:-

-- Executing AGI ("SIP/801-09467e70", "agi-//127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577//call_log completed, returning 0
-- Executing Dial("SIP/801-09467e70", "sip/12146354605@SIPtrunk|30|o") in new stack
-- Called 12146354605@SIPtrunk
-- Nobody picked up in 3000 ms
-- Executing Hungup("SIP/801-09467e70","") in new stack
== Spawn extension (default, 12146354605, 3) exited non-zero on 'SIP/801-09467e70'
-- Exexcuting DeadAGI("SIP/801-09467e70", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log comleted, returning 0
-- Executing DeadAGI("SIP/801-09467e70", "agi://127.0.0.1:4577/VD_hangup--HVcauses-PRI-----NODEBUG---------16-----NOANSWER------------)"0 in new stack
-- AGI Script agi://127.0.0.1:4577/VD_hangup--HVcause ... ----------) completed, returning 0

sounds like an asterisk question to me .... i would suggest asterisk forum for such questions ...
you have to be able to call from asterisk before you go on to configure vicidial & not viceversa.

-- Executing Dial("SIP/801-09467e70", "sip/12146354605@SIPtrunk|30|o") in new stack
-- Called 12146354605@SIPtrunk
-- Nobody picked up in 3000 ms

your configuration is working
call ended 30sec, as per the settings.


try to call other number and see.

rather calling only one number

ram

two visible arrors

1)

[SIPtrunk]
type=friend
host=VoIPProvider's_IP_Add
fromdomain=VoIPProvider's_IP_Add
fromuser=almtech
secret=my_password
username=my_username
context=default
authname=my_username
;dtmfmode=inband
canreinvite=yes
qualify=no
nat=yess ----------------------------------

2)


Example phone extensions

; dial a long distance outbound number through a SIP provider
exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _1NXXNXXXXXX,2,Dial(sip/${EXTEN}@SIPtrunk,30,o)
exten => _1NXXNXXXXXX,3,Hangup


use this
comment the exten => _1NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
Example phone extensions

; dial a long distance outbound number through a SIP provider
exten => _1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _1NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _1NXXNXXXXXX,2,Dial(sip/${EXTEN}@SIPtrunk,30,o)
exten => _1NXXNXXXXXX,3,Hangup


and remember to use the sip-silence config for the 8365 context


and what country are you living??????????????????????????

that block the 5060 port