I have a DID with Anveo direct, and I want to have inbound calling only with my Asterisk server. I have confirmed everything is working using sip2sip (I call using Sip2Sip registered on Zoiper on my cell phone; and Sip2Sip registered on Asterisk with pjsip.conf). Now I have a DID with AnveoDirect, and am of the understanding that I do not register with Anveo and pjsip.conf. Here’s what I have done so far:
My Asterisk server has an A name, and can be found at my.domain.com.
In Anveo, Inbound Service > Configure Destination Trunks > Add a New SIP Trunk > email@example.com (obviously 15551235555 being my DID)
Inbound Service > Configure DIDs > edit > Call Options > And then I choose my destintion SIP trunk that I created (and I also checked off G.711u for codecs)
Note: I have also tried $[E164]$@my.domain.com and it also doesn’t work.
In Asterisk pjsip.conf is where I get lost and feel like I’m guessing. Here’s what I have in pjsip.conf (based off of this post):
[transport-udp] type=transport protocol=udp bind=0.0.0.0:5060 [siptrink-aor] type=aor contact=sip:my.domain.com [siptrunk] type=endpoint transport=transport-udp context=from-siptrunk disallow=all allow=ulaw aors=siptrunk-aor send_diversion=yes send_pai=yes send_rpid=yes trust_id_inbound=yes direct_media=no rtp_symmetric=yes force_rport=yes rewrite_contact=yes [siptrunk-identify] type=identify match=my.domain.com endpoint=siptrunk
I see that the AnveoDirect faq also suggests sbc.anveo.com but that’s for sip.conf. I tried changing contact and match to sbc.anveo.com and mixing and matching different combinations, but still can’t get it to work. When I call my DID, I get a busy signal.
Also, I set up iptables to secure my asterisk server (because it’s on GCP compute) using this tutorial. For the tutorial, I changed
YOUR_HOSTNAME.no-ip.com to my.domain.com (obviously my.domain.com being my actual domain).
It’s probably something really simple, but I’m very new to Asterisk and not understanding how to receive calls.
Edit more information. I realize now that I had gotten rid of the siptrunk-registration section in my old pjsip.conf:
[siptrunk-registration] type=registration transport=transport-udp outbound_auth=siptrunk-auth server_uri=sip:sip2sip.info client_uri=sip:firstname.lastname@example.org contact_user=inbound-calls retry_interval=60
I got rid of it because I don’t need to register the AnveoDirect DID? But the problem now is that it had
contact_user=inbound-calls which is then used in extensions.conf:
[from-siptrunk] exten => inbound-calls,1,Verbose(1,Playing some music.) same => n,Answer same => n,MusicOnHold(ulawstream) same => n,Hangup()
So I don’t really know where to go from here.