Does the new T-mobile KVD21 block ports 5060 and RTP?

[pugetnet]

I have exaused all options to make both my polycome, my soft-phone and now my asterisk box to register with my carrier in Canada. It has 1000,000 subscribers. I have heard that in the past this modem did work, but the company may have disabled the port. The modem web interface has ZERO options to set for the modem. I need to know if I need the commercial version and can I used any LTE modem of my choosing? Pep-wave or Cradle Point comes to mind?
I was also told to enable LTS and it will work but really? port scan shows 5060 closed!

Here is the ver I am using! Asterisk 13.18.3~dfsg-1ubuntu4

sip show peers default/374848_server (Unspecified) Auto (No) No 0 Unmonitored

sip show registry
mysipservice:5060 N myusername 105 Registered Thu, 05 Dec 2024 15:47:32
1 SIP registrations.

my sip.conf

making a call CLI feed back
Call from ‘200’ (192.168.12.5:5060) to extension ‘125xxxxxxx’ rejected because extension not found in context ‘default’.

Sip settings in sip.conf
[general]
context=public ; Default context for incoming calls. Defaults to ‘default’
register =>374848_server:I123454321i@seattle.voip.ms:5060/default
allow=ulaw
allow=alaw
allow=g722
allow=g729

[default]
type=friend
username=374848_server
secret=I123454321i ; Make sure this is correct
dtmfmode=rfc2833
;host=seattle.voip.ms
context=inbound
canreinvite=no
fromuser=200
register=yes
fromdomain=seattle.voip.ms
register_timeout=5

[200]
type=friend
username=200 ; The username or extension number
secret=123454321 ; The SIP password
host=dynamic ; The host can be dynamic if the device gets its IP from DHCP
context=default ; The dialplan context to use
callerid=“Polycom 200” <200>
;codec=alaw,g729,gsm,opus,pcmu,pcma,g722 ; Audio codecs
nat=yes ; If the device is behind NAT
qualify=yes ; Keep track of device status
dtmfmode=rfc2833 ; DTMF signaling method

BTW I am really rusty!!! its been years since setting up a asterisk box.

this line is in the [default] context in extensions
exten => NXXNXXXXXXX,1,Dial(SIP/seattle.voip.ms/${EXTEN})

I also have in sip.conf
[general]
context=public
register => username:password@seattle.voip.ms:5060/default

[pugetnet]

I have worked really hard to resolve this issue and I don’t think short of calling t-mobile management, why sip traffic wont pass over port 5060. I worked on instead using port 5061 with the TLS and the keys it reads. I don’t see the errors on the screen anymore “Wrong path to the keys” for openssl. I do a sip show registry and it shows my account registered with the VoIP company. But sip show peers shows not registered. I can telnet into the VoIP account and do other test based on what chatgtp has shown. So it looks like it can connect, but I think its only one way communications.

[david551]

chan_sip is no longer supported

Allowing all codecs, as you are doing, by not disallowing any, can produce over long request packets that can get truncated and cause problems.

Your default context doesn’t seem to be used, and I think register has to be in general.

type=friend is over used (may not be needed at all).

Your exten definition is not in the pattern match format (and if it were, you have redacted one of the fourth digit, so we can’t confirm it isn’t zero.

[pugetnet]

you mean asterisk dev does not support chan_sip? then what is supported?
The errors are gone. at least from the CLI.
type=fried is over used ? can you explain?
Extension definition? is not in the pattern match format?

[david551]

chan_pjsip.

type=friend is a combination of type=peer and type=user. In most cases you only need the type=peer part and using type=friend can cause misoperation and security issues, Typically providers won’t authenticate themselves, so you have to use remotesecret or insecure=inivite. If you make them type=friend, anyone knowing the peer name you have used can spoof calls as arriving from them. type=friend on phones often causes problems when people have a tie trunk and the other side has overlapping extension numbers.

Dialplan patterns are introduced by an initial underscore. Without that they have to match character for character. X and N are valid characters in SIP user names.

[pugetnet]

I will need to make the habit of backing up sip and extensions lines every time I make a change. The extnsions 200 and 201 are identical. pressing 8500 for voicemail main works on the softphone. The hard polycom phone shows the steps that the the polycome phone are going though on cli…but no audio output on speaker. Yesterday I did have it working.

[pugetnet]

The phone does not register so will be finding another provider.