vicidial.org

Hello again,

I was wondering how to configure Goautodial to call international numbers, there are numbers I would like to call within Africa, Europe, Asia & South America

My dialplan entry is:
exten => _9011X.,1,AGI(agi://127.0.0.:4577/call_log)
exten => _9011X.,n,Dial(SIP/INTLDialer/${EXTEN:4})
exten => _9011X.,n,Hangup

is there a way to configure this?


GoAutoDial CE 3.0 | Asterisk 1.8.2 | Linux VPS Server | No Digium/Sangoma Hardware (but DAHDI is installed already on the server) | No Extra Software After Installation

Hi Good day!

To be able to do this you need to check the dial prefix to another country, then you just need to add it on your dialplan. For example:

44XXXXXXX. - uk
61XXXXXXX. - australia
81XXXXXXX. - japan
1XXXXXXXX. - canada
65XXXXXXX. - singapore
66XXXXXXX. - thailand
852XXXXXX. - hongkong

Hope this helps! Cheers!

so if I want to dial to Japan I should configure it like this?

exten => _901181XXXXXXX.,1,AGI(agi://127.0.0.:4577/call_log)
exten => _901181XXXXXXX.,n,Dial(SIP/INTLDialer/${EXTEN:1})
exten => _901181XXXXXXX.,n,Hangup

I did the configurations, but I get a busy/congested error

[Mar 7 16:47:46] -- Executing [011216210040528@default:2] Dial("SIP/8001-00000092", "SIP/INTLDialer/11216210040528") in new stack
[Mar 7 16:47:46] == Using SIP RTP CoS mark 5
[Mar 7 16:47:46] WARNING[23244]: chan_sip.c:6033 sip_call: No audio format found to offer. Cancelling call to 11216210040528
[Mar 7 16:47:46] -- Couldn't call SIP/INTLDialer/11216210040528
[Mar 7 16:47:46] == Everyone is busy/congested at this time (0:0/0/0)
[Mar 7 16:47:46] -- Executing [011216210040528@default:3] Hangup("SIP/8001-00000092", "") in new stack
[Mar 7 16:47:46] == Spawn extension (default, 011216210040528, 3) exited non-zero on 'SIP/8001-00000092'
[Mar 7 16:47:46] -- Executing [h@default:1] AGI("SIP/8001-00000092", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL----------") in new stack
[Mar 7 16:47:46] -- <SIP/8001-00000092>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

Hi, have you input your SIP details for your VOIP correctly? You should ask this on your provider if your calls were receive by them.

When you mean SIP details is that something I should add within sip.conf?

they said they didn't see any calls coming though, what is wrong?

Hi Good day!

Since the calls were not going in their server you can coordinate the SIP details that your provider will set and let them authenticate your public IP being used by your dialer. And if it is behind firewall just enable SIP ports 5060-5070.

thanks for your reply I switched codecs now the calls get established but there is no audio
what should I do?