Outgoing call via SIP trunk problem

Asterisk Asterisk SIP
1

Hello guys,
I will appreciate any help. My problem:

I trying to connect to my VoIP provider. Registration seems to be OK, Incoming calls work fine too. But outgoing calls does not.

My setting in sip.conf:

register => username:password@provider.domain/provider's_number

[random_name]
type=peer
username=username
secret=password
host=provider.domain
port=5060
context=From-VoIP
insecure=port,invite
canreinvite=no
disallow=all
allow=alaw
bindport=5060
qualify=yes
qualifyfreq=60

extensions.conf:
[out-to-international]
exten => _X., 1, Dial(SIP/random_name/${EXTEN})
   - There I also tried: 
   exten => _X., 1, Dial(SIP/${EXTEN}@random_name)
 - And also almost every combination

But the result is always same:

NOTICE[2758]: chan_sip.c:21337 handle_response_invite: Failed to authenticate on INVITE to '"My server" <sip:100@IP_of_my_PBX>;tag=as6a9d2792'
    -- SIP/random_name-00000063 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'SIP/100-00000062' status is 'CONGESTION'

In wireshark i see this message:
407 proxy Authentication message Required, so I send INVITE with Proxy-Authorization, but after that I got 503 Service unaviable response.

So I’m obviously doing something wrong with registration, but I can not figure out what. How Proxy-Authorization header should looks like? Especialy Authentication uri.

I’ve got:
“sip:username@provider.domain” or “sip:provider’s_number@provider.domain” or “sip:number_which_I_calling@provider.domain”
depends on how I configure Dial func in extension.conf

Thanks for any reply, I’m lost with that

2

Firstly, you should not be persisting in trying to make an unsupported driver that will be completely removed in 2023 work; you should be using chan_pjsip.

My best guess is that it wants fromuser setting.

username is essentially useless if you are doing the registering.

Although you will need insecure=invite (although remotesecret is a better approach), you are unlikely to need insecure=port.

canreinvite was renamed to directmedia about a decade ago

You haven’t configured an authorisation section, so I wouldn’t expect proxy authentication to work, except maybe by treating it like user agent authentication.

3

Thank you for reply,
I know that about pjsip and in common mounths I’d like to use it in newer systems.

Thank you for answer,
I know about pjsip and in the following months I would like to use it in newer systems.

Unfortunately provider told me I should not use fromuser.
I changed insecure to invite only

And that is maybe the problem - configuration of authorisation section. But can you be more specific? How should I configure it?

and also: when I run sip reload I always see the line:

doing dnsmgr_lookup for ‘provider.domain’
So it look like the DNS translation does not work am I right? How can I fix this?

Thanks for advices I’m really frustrated about this problem with this concrete trunk

4

It’s actually authentication. I was going from memory.

https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L1214

Details depend on your provider, and your account with them.

5

Thank you David!
I’ve never seen that before (sad sigh).
But it looks like exactly same as register string. So I filled with same data as register string and Whoohuuuu it’s working!!

So thank you very much for solving this! It took me few long night

6

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