Asterisk and Skyetel SIP Issues

Hello!
I’m currently building a system that uses Asterisk with Skyetel, but I’ve run into a roadblock due to Skyetel’s use of dynamic IP addresses for their SIP endpoints.

Here is my current PJSIP config:
`[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060

[skyetel_auth]
type=auth
auth_type=userpass
username=USERNAME
password=PASSWORD

[skyetel_aor]
type=aor
contact=sip:nw1.reg.skyetel.com
qualify_frequency=60

[skyetel_endpoint]
type=endpoint
aors=skyetel_aor
auth=skyetel_auth
context=MY_CONTEXT
disallow=all
allow=ulaw
outbound_auth=skyetel_auth
from_user=USERNAME
from_domain=nw1.reg.skyetel.com
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
transport=transport-udp
identify_by=username
direct_media=no

[anonymous]
type=endpoint
context=MY_CONTEXT
disallow=all
allow=ulaw
transport=transport-udp
direct_media=no
trust_id_inbound=yes
send_pai=yes
send_rpid=yes
rtp_symmetric=yes
force_rport=yes

[skyetel_registration]
type=registration
outbound_auth=skyetel_auth
server_uri=sip:nw1.reg.skyetel.com
client_uri=sip:USERNAME@nw1.reg.skyetel.com
contact_user=USERNAME
retry_interval=60
forbidden_retry_interval=300
expiration=3600
transport=transport-udp`

The issue is that without the anonymous endpoint, inbound calls fail because Asterisk doesn’t recognize the source IP (which is always changing). Skyetel doesn’t provide a fixed IP range, so I can’t rely on the identify sections based on IP matching either.

I’ve checked Skyetel’s documentation, but it appears outdated and doesn’t reflect their current dynamic IP behavior, especially for pjsip. I’m hoping someone has a workaround or a way for handling this setup.

I am running Asterisk version 20.6.0~dfsg+~cs6.13.40431414-2build5 on Linux Ubuntu 24.04.2 LTS.

I’m still new to Asterisk, so any guidance would be greatly appreciated.

Thanks!

You are missing a type=identify section, which, unlike chan_sip, can be configured to match all the address ranges they use.

I actually did try that before. However, since skyetel uses dynamic IP addresses for their SIP traffic and I haven’t been able to find a list of IPs or ranges they use, the type=identify section doesn’t work, which is why I had resorted to using anonymous just to get inbound calls working. What I had tried to do was call my asterisk system, get the address from the 401 error, log it. Every time I did it, it was a different address and after about 50 calls of trying, I gave up.

The provider should be able to tell you the range. Anything is i likely to be better than 0/0.