Goautodial 4 + SIP Gateway
Added by nik Papa over 5 years ago
Hello everyone,
first of all thanks and congrats on your amazing work.
I managed to install goautodial 4 on my Server with a static IP.
I am at the state where I need to set up a carrier and make a manual call from the browser, or even an automated call through a campaign.
The Carrier settings :
[XXXX]
type=peer
host=192.168.x.x
port=5060
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=g729
allow=ulaw
qualify=yes
insecure=port,invite
The dialplan entry:
SIPTrunk = SIP/XXXX
exten => _39X.,1,AGI
exten => _39X.,2,Dial(SIPTrunk/@{EXTEN},,tTo)
exten => _39X.,3,Hangup()
Protocol SIP
output of asterisk: sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
XXXX 192.168.x.x Yes Yes 5060 OK (5 ms)
kamailio 127.0.0.1 Yes Yes 5060 OK (1 ms)
Where XXXX is my gateway. The device that is connected to a PRI line and handles the outbound calls.
I can log in as an agent, and try to make a call through the dialer.
However the answer I get is call timeout and I do not know where to begin searching.
Can anyone give me a hint which logs to look at, as it seems there's no logging taking place while an agent makes the call.
Any help/guidance would be much appreciated I can provide any info you need.
Thanks in advance!
Replies (8)
RE: Goautodial 4 + SIP Gateway
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Added by Vericar De Espiritu Santo over 5 years ago
Sorry to answer, but can you tellme how do you could config the Dialer, my GOautodial keeps freeze after select campaing, and i cant pass it. can you help me?
RE: Goautodial 4 + SIP Gateway
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Added by nik Papa over 5 years ago
Hey there, I had to make sure kamailio was running and that I used Firefox
Check the browser console as well as the php logs of your server. Make sure that kamailio is an asterisk peer and be sure that kamailio can connect to the database.
To see the peers
Asterisk -r and sip show peers
If there's anything else I can help you with, don't hesitate to ask.
RE: Goautodial 4 + SIP Gateway
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Added by giorgio gio about 5 years ago
Ciao Nik. Sei riuscito a risolvere il problema?
RE: Goautodial 4 + SIP Gateway
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Added by Enzo Zazzaro about 5 years ago
ciao risolvi cosi in CARRIER
[SIP]
disallow=all
allow=gsm
allow=g729
allow=ulaw
type=friend
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=invite,port
nat=yes
host=il tuo provideer
username= nome utente
secret= passowrd
allow=alaw
e sotto
exten => _39XX.,1,AGI
exten => _39XX.,2,Set(_Missing_CLID1=${CALLERID})
exten => _39XX.,3,Dial(SIP/${EXTEN}@SIP,,tTo)
exten => _39XX.,4,Hangup
e vedi che fai le chiamate
RE: Goautodial 4 + SIP Gateway
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Added by giorgio gio about 5 years ago
Enzo Zazzaro wrote:
ciao risolvi cosi in CARRIER
[SIP]
disallow=all
allow=gsm
allow=g729
allow=ulaw
type=friend
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=invite,port
nat=yes
host=il tuo provideer
username= nome utente
secret= passowrd
allow=alawe sotto
exten => _39XX.,1,AGI
exten => _39XX.,2,Set(_Missing_CLID1=${CALLERID})
exten => _39XX.,3,Dial(SIP/${EXTEN}@SIP,,tTo)
exten => _39XX.,4,Hangupe vedi che fai le chiamate
Grazie Nick per la risposta.
Come carrier adopero quello che ho su un'altro server con la versione di Goautodial 3.3 e funziona perfettamente.
Non so perchè ho problemi con la WebRTC
Non capisco perchè non si allinea. Un volta fatto il log non si sente neanche la voce di allineamento.
Puoi darmi qualche dritta?
Grazie davvero
RE: Goautodial 4 + SIP Gateway
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Added by Wittie Manansala about 5 years ago
Hi,
Please provide us the following for us to review:
1. nano /var/www/html/php/Config.php
2. nano /var/www/html/php/goCRMAPISettings.php
3. nano /etc/kamailio/kamailio.cfg
4. nano /etc/rtpengine/rtpengine.conf
5. nano /etc/asterisk/sip-goautodial.conf
6. systemctl status kamailio
7. systemctl status ngcp-rtpengine or systemctl status rtpengine
Reference Link:
https://goautodial.org/projects/goautodialce/wiki/Goautodial_Getting_Started_Guidev4
Thanks
Thanks
RE: Goautodial 4 + SIP Gateway
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Added by Rodolfo Josue Vallecillo Ramirez over 3 years ago
giorgio gio wrote in RE: Goautodial 4 + SIP Gateway:
Enzo Zazzaro wrote:
ciao risolvi cosi in CARRIER
[SIP]
disallow=all
allow=gsm
allow=g729
allow=ulaw
type=friend
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=invite,port
nat=yes
host=il tuo provideer
username= nome utente
secret= passowrd
allow=alawe sotto
exten => _39XX.,1,AGI
exten => _39XX.,2,Set(_Missing_CLID1=${CALLERID})
exten => _39XX.,3,Dial(SIP/${EXTEN}@SIP,,tTo)
exten => _39XX.,4,Hangupe vedi che fai le chiamate
Grazie Nick per la risposta.
Come carrier adopero quello che ho su un'altro server con la versione di Goautodial 3.3 e funziona perfettamente.
Non so perchè ho problemi con la WebRTC
Non capisco perchè non si allinea. Un volta fatto il log non si sente neanche la voce di allineamento.
Puoi darmi qualche dritta?Grazie davvero
Hola he presentado el mismo problema, lo que me ha llevado a realizar muchas pruebas, teniendo como resultado que el servicio kamailio solo esta realizando el alineamiento por 127.0.0.1, si logran configurarlo por otra ip les agradezco me hagan saber tengo el mismo problema