I have registered with telnyx but i couldnt make any calls using us number incoming and outgoing can anyone help what i am missing?

Asterisk Asterisk SIP
1

this is my pjsip.conf where i am configuring with telnyx

i have registered with telnyx but i couldnt make any calls using us number incoming and outgoing can anyone help what i am missing ?

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

[telnyx_registration]
type=registration
outbound_auth=telnyx_auth ; Reference correct auth section here
server_uri=sip:xxxxxxx:5060
client_uri=sip:xxxxxx@xxxxxxxx:5060
retry_interval=60

[telnyx_auth]
type=auth
auth_type=userpass
username=xxxxxx ;
password=xxxxxxx ;

[telnyx_aor]
type=aor
contact=sip:xxxxxxxxxx:5060

[telnyx_endpoint]
type=endpoint
context=from-pstn ;
disallow=all
allow=ulaw,alaw,g729,g722
aors=telnyx_aor

[telnyx_identify]
type=identify
endpoint=telnyx_endpoint
match=sip.telnyx.com

[1001]
type=endpoint
context=from-internal
disallow=all
allow=ulaw,alaw,g729,g722
dtmf_mode=rfc4733
rewrite_contact=yes
force_rport=yes
aors=1001_aor
auth=auth_1001
callerid=“Test User” <1001>

[1001_aor]
type=aor
max_contacts=1
remove_existing=yes
minimum_expiration=30

[auth_1001]
type=auth
auth_type=userpass
username=1001
password=xxxxxxxxxx

stextensions.conf

cat extensions.conf.telnyx
[from-pstn]
; Handle incoming calls from PSTN
exten => _+1NXXXXXXXXX,1,NoOp(Incoming call from PSTN to +1NXXXXXXXXX)
same => n,Dial(PJSIP/1001)
same => n,Hangup()

exten => _NXXXXXXXXX,1,NoOp(Incoming call from PSTN to NXXXXXXXXX)
same => n,Dial(PJSIP/1001)
same => n,Hangup()

[from-internal]
; Dial 10-digit US number - prepend +1 and send to Telnyx trunk
exten => _NXXNXXXXXX,1,NoOp(Outbound call to ${EXTEN})
same => n,Dial(PJSIP/+1${EXTEN}@telnyx_endpoint) ; Updated endpoint reference
same => n,Hangup()

; Catch-all rule for other number formats (e.g., international)
exten => _X.,1,NoOp(Outbound call to ${EXTEN})
same => n,Dial(PJSIP/+${EXTEN}@telnyx_endpoint) ; Updated endpoint reference
same => n,Hangup()

2

You haven’t specified what actually happens. You would need to show Asterisk console output, as well as SIP traffic using “pjsip set logger on”.

3

You either haven’t consistently redacted the same domain name, or sip.telnyx.com should appear in, at least, the contact= line and the server_uri= line. Note that we have to assume the last two are different, as they have been redacted differently. That is unusual.

It’s also possible that they use SRV records, in which case you shouldn’t be specifying a port.