SIP retransmissions (firewall and NAT)

Asterisk Asterisk Support
1

When using telnyx for siptrunking outbound calls work no problem but inbound calls never even show in Asterisk:

telnyx:

Using siptrunk for sip trunking, inbound and outbound calls fail, but do show:

siptrunk:

The particular error for inbound calls with siptrunk references:

https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

which mentions NAT and firewalls as being potential problems. Because I can dial out with telnyx does that mean that I can eliminate NAT and firewall? I ran online port scanners with sipscanner.voicefraud.com and it found:

123.456.789.123 5060 Linksys/SPA942-6.1.5(a)

What can I do to test NAT and firewall? Is there a positive test to just say “NAT works” or “no firewall problems”?

A failed call to siptrunk:

2

I am not sure what you are trying to show with all this apparently unrelated information. The basic problem is that inbound does not work correct?

Not seeing anything in Asterisk CLI often means your host= or context= is wrong. Sometimes you still see something in Asterisk CLI but sometimes not. Also make sure Asterisk is listening on the SIP port the provider is sending to.

You should probably post your network information and asterisk configs instead.

So provide basic network info (Asterisk and extensions behind NAT?), your asterisk trunk configuration and sip_nat.conf.

Also try capture SIP with tcpdump from the provider IP while attempting an inbound call and post the results here.

tcpdump -nqt -s 0 -A host xx.xx.xx.xx and port 5060

If your asterisk is behind NAT I am assuming you are port forwarding 5060? Also disable SIP ALG if enabled.

3

I did, briefly, have both inbound and outbound context’s working. However, I’ll need asterisk 24x7 so went ahead and put it on a droplet with digital ocean.

My router forwards 5060-5061 UDP and 10000-20000 UDP, yes, SIP ALG is disabled. It was forwarding to the Asterisk box, but now forwards to the IP phone.

4

You don’t need the port forward. Also check if you have enabled the in your box.
Behind a router is needed but on a public server no.

5

Changed nat=none since Asterisk is running on a public server.

On my localnet, I have port forwarding to the IP address for the hardphone.

You wrote “check if you have enabled the ______ in your box” – what’s the _____? NAT?

for each entry in sip.conf I specified the external IP for my home address.

Is it a problem that the internal ip for thufir shows below, or is that ok?

<--- SIP read from UDP:<my.external.ip>:5060 --->
SIP/2.0 404 Not Found
To: <sip:thufir@192.168.1.5:5062>;tag=a969ad7ba51714ai0
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as7b5d6e61
Call-ID: 665575bf64e656f0541937e6386a78db@<asterisk.droplet.ip>:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP <asterisk.droplet.ip>:5060;branch=z9hG4bK4b216d56
Server: Linksys/SPA942-6.1.5(a)
Content-Length: 0

full trace: