Inbound external calls not registering on logs/debug

Asterisk Asterisk SIP
1

Hi, sorry if I miss some information, I am kind of new to this, and wanted to set up an automatic cellphone answer. My problem came after trying to set up the sip number i got via an sip provider (iptel). I set everything up, and it appears as registered, but the call I make from my cellphone does not register.

SIP.CONF:
[general]
port=5060
bindaddr=0.0.0.0
context=public
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=yes
qualify=yes
nat=force_rport,comedia
canreinvite=no
dtmfmode=rfc2833
srvlookup=yes

register => 11111111:password@ser.ver.com

[provider]
type=peer
context=provider
allow=ulaw,alaw
qualify=yes
username=11111111
secret=password
host=ser.ver.com
nat=force_rport,comedia
insecure=port,invite
fromdomain=ser.ver.com
defaultuser=11111111
authuser=11111111
fromuser=11111111
dtmfmodf=rfc2833

EXTENSIONS.CONF

[provider]
exten => _X.,1,Answer()
exten => _X.,n,Wait(10)
exten => _X.,n,Hangup()

sip show registry
Host dnsmgr Username Refresh State Reg.Time
ser-ver.com:5060 N 11111111 585 Registered Fri, 20 Sep 2024 18:30:11
1 SIP registrations.

sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
provider/11111111 190.1.1.1(changed ip) Yes Yes 5060 OK (32 ms)

I turned on sip debug, and also get the ocurrent transaction for registering

2

Why are you using the unsupported chan_sip driver? This is not even present in the latest two versions of Asterisk.

Are you behind NAT, as nothing in your configuration deals with that. (nat= is for when the peer is behind NAT and fails to compensate for that; the default works in most cases).

defaultuser and username are the same option and neither of them are used when you have a named host.

Specifying allow without disallow can cause problems, because packets get too large.

dtmfmodf is not a valid parameter name.

3

I was following a document the sip provider gave me, but they aren’t answering to my questions anymore.

I was indeed behind a NAT, what should I do to allow the connection for the call?
$ traceroute PUBLICIP
traceroute to PUBLICIP (PUBLICIP), 30 hops max, 60 byte packets
1 _gateway (192.168.0.1) 13.415 ms 13.387 ms 13.367 ms
2 10.54.216.1 (10.1.1.1) 24.660 ms 24.571 ms 24.582 ms
3 100.72.7.185 (100.1.1.1) 24.578 ms 31.564 ms 31.589 ms
4 192.168.65.185 (192.1.1.1) 31.624 ms 31.580 ms 31.580 ms

Thanks for the detail about defaultuser and username, deleted them.
Also added disallow=all above the allows, dtmfmodf was a typo, I changed it to dtmfmode

4

https://docs.asterisk.org/Asterisk_21_Documentation/API_Documentation/Module_Configuration/res_pjsip/

Also external_signalling_address

5

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