Zoiper not connecting

Hi, I am brand new to all of this, and I’m just playing around trying to learn. So, apologies for stupid questions/ridiculous notions in advance.

I’m running asterisk in an AWS Ubuntu server. I have Zoiper installed on my local PC and it’s not connecting SIP UDP, or anything else for that matter. I’m also using GoTrunk.

This is what I have keyed into my sip.conf:

[general]
allowguest=no
context=default
bindport=5060
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g723
allow=g726
allow=speex

; replace INBOUND_SIP_USERNAME and INBOUND_SIP_PASSWORD
; with your Inbound SIP Registration credentials
register => xxxx:xxxx

[201]
type=friend
host=dynamic
context=from-internal
username=201
password=xxxxx
qualify=yes
nat=force_rport,comedia

[202]
type=friend
host=dynamic
context=from-internal
username=202
password=xxxx
qualify=yes
nat=force_rport,comedia

[trunk]
type=peer
host=eu.st.ssl7.net ; Europe POP
;host=amn.st.ssl7.net ; North America POP
context=from-trunk
qualify=yes
defaultuser=xxxx
remotesecret=xxxxx

I am also unsure of which IP address to key into Zoiper. I’m assuming the public IP?

Thanks!

In that case, you should be using chan_pjsip, not the deprecated and unsupported chan_sip.

The address in zoiper should be the address of Asterisk as seen by zoiper, which would normally be the LAN address, or even 127.0.0.1, if is on the same machine. I would advise against starting with it on the same machine, as that introduce additional complications.

You should start with echo and playback applications, rather than going straight to an outgoing call.

Only use type=friend if you have more than one phone with the same address (but chan_pjsip is rather different, so don’t bother learning chan_sip).

The register line looks incomplete.

defaultuser is not useful for a non-dynamic peer.

username is the obsolete name for defaultuser, is rarely useful, and then only with a default domain.

It is not clear why nat= is not left at default, although it would be unusual for it to cause any harm.

Sorry, I should have added you need to provide logging showing requests reaching Asterisk and how they failed.

Also, unless the public address you mention is directly on the Asterisk machine, you will need to tell Asterisk what it is and also which networks should be treated as not using it.

Just FYI, I have been successfully doing some SIP testing using Twinkle as the SIP-phone client, connecting to an Asterisk server on the same machine. I keep them out of each other’s hair by letting Twinkle use port 5060, and putting Asterisk on another port, e.g. 506.

This way I make my embarrassing mistakes on my own private LAN. :wink: