Configurazione Trunk Irideos

Buongiorno a tutti,
sperando di fare cosa gradita indico qui di seguito la conifgurazione del trunk SIP in caso di full-voip con Irideos. Ci ho sbattuto la testa per una settimana senza successo con PJSIP ma alla fine la configurazione seguente in un trunk chan_sip funziona all’istante:

Outgoing Peer Details
type=peer
insecure=port,invite
contex=inbound-trunk ;context for incoming call
srvlookup=no
host=vnpublic.irideos.it
port=5060
transport=udp
fromdomain=vncluster.irideos.it
username=user fornito da Irideos
secret=password fornita da Irideos
canreinvite=no
nat=no
disallow=all
allow=g729
allow=alaw
qualify=yes
dtmfmode=rfc2833
dnisontoheader=no
defaultexpiry=120
minexpiry=60
maxexpiry=3600

Incoming Register string
user fornito da Irideos@vncluster.irideos.it:password fornita da Irideos@[vnpublic.irideos.it:5060/user fornito da Irideos](http://vnpublic.irideos.it:5060/***user fornito da Irideos***)

Please try:

Username: user fornito da Irideos
Secret: password fornita da Irideos
Authentication: Outbound
Registration: Send
SIP Server: vncluster.irideos.it
Expiration: 120
Outbound Proxy: sip:vnpublic.irideos.it\;lr\;hide
Contact User: user fornito da Irideos
From Domain: vncluster.irideos.it
From User: user fornito da Irideos (not shown in your chan_sip but I suspect is needed)

On Codecs tab, check only alaw and g729, in that order.

FreePBX does not have a built-in context named inbound-trunk. If you have built that and it’s working properly, also set:

Context: inbound-trunk

If you have trouble and the trunk doesn’t register, paste the Asterisk log for a failed registration attempt, with pjsip logger on, at pastebin.com and post the link here.

If registration is ok but you can’t receive calls, paste the log for an incoming call attempt.

If outbound calls fail, paste the log for a failing outbound call attempt.

If accurate, the OPs configuration doesn’t set a context, as the keyword is incorrect.

The following could cause problems if correct and actually needed.

An @ in a SIP username isn’t valid in SIP. Reportedly some ITSPs still want it. It is valid if encoded as %40, but if the provider requires it and requires it literally, I’m not sure if and how to get that past validation.

Hi!
I apologize to both of you if I caused any confusion (especially writing in Italian!).
My post is not for a problem but is to try to solve it for others. In simple words, that is the working configuration to use the Full-Voip Trunk with the Irideos operator in Italy.
Both incoming and outgoing calls work complete with CID and DTMF functionality (both incoming and outgoing)

@Stewart1 yet it works like this without any problems (I don’t think I created the indicated context)

@david55 this is what is required by the ISP (and work…)

Sorry again for my misunderstanding!

If Stewart’s solution worked, it was not required by the service provider. as he ignored the first domain entirely.