Unable to receive incoming calls on a trunk with outboundproxy

Asterisk Asterisk SIP
1

Hello guys,

I am running Asterisk 13.35.0. I have configured a SIP account as a Trunk, because I want to use it’s DID number.

The settings from the provider look like this:
Username: some-username
Password: some-password@
Host: domain.com
Outbound Proxy: sip.domain.com
Port: 5060
Fromuser: I don’t know if its correct, but I put some-username
Fromdomain: I don’t know if its correct, but I put domain.com

The register string looks like this:
some-username@domain.com:some-password@@sip.domain.com:5060/some-username

Config looks like this

[trunk-name]
user=some-username
username=some-username
secret=some-password@
disallow=all
allow=g729,gsm,opus,alaw,ulaw
directmedia=no
context=billing
dtmfmode=RFC2833
insecure=port,invite
nat=force_rport,comedia
qualify=yes
type=peer
host=domain.com
fromdomain=domain.com
fromuser=some-username
sendrpid=no
outboundproxy = sip.domain.com

[trunk-name1](trunk-name);
host=the-ip-of-domain.com (calls appear to come from it, the IPs of proxy and domain are different)

sip show registry says this:

sip.domain.com:5060                     N      some-usernam       105 Registered           Sun, 20 Mar 2022 16:05:11

However, when I try to call the DID number, I get an error:

[2022-03-20 16:06:38] NOTICE[29542][C-00002e47] chan_sip.c: Failed to authenticate device <sip:MY-CALLER-PHONE-NUMBER@domain.com>;tag=6524E8C1-623742FE000016EC-0A009700 for INVITE, code = -1

sngrep reports a 401 unauthorized and a 403 forbidden error code for the invite string.

I didn’t test outbound calls yet. Any ideas what to do?

My regular trunks work perfectly fine. The difference with them is the presence of proxy.

P.S. I know Asterisk 13 is EOL and using chan_sip seems frowned upon here, but I can’t really upgrade at the moment/change the software stack, as it would require a significant investment and too much time. If you can’t/don’t want to help me due to this fact that’s completely fine, I still appreciate your time reading.
Thanks.

2

Tip to troubleshooting, try compare the authentication in your REGISTRE with your INVITE

3

Yes, the invite is not containing my username. They send the header with caller-id@domain.com . I unfortunately cannot remedy this - it is just how their system works.

4

I suspect they are not sending from the first DNS record’s for domain.com (I’m guessing that domain.com was really supposed to be example.com, although I never understand why people obfuscate the ITSP, as the ITSP name is important information).

The correct solution to this is to use chan_pjsip and include match lines for all the possible addresses and address ranges from which they can send.

5

The ITSP is vmobile.eu - it is a local provider. Is there no way to get this working with chan_sip?

P.S. I forgot to say that it claims the “trunk-name1” peer unreachable. I guess this is because registration requires the proxy to work. “trunk-name” is fine.

6

The workarounds for chan_sip are to either create peer entries for very address they use, or to enable allowguest.

7

Thanks. I have changed host on

[trunk-name1](trunk-name);
host=the-ip-of-domain.com

to

[trunk-name1](trunk-name);
host=the-ip-of-sip.domain.com

and now both peers show reachable.

However I get a new error:

[2022-03-20 22:56:43] NOTICE[29542][C-000033eb] chan_sip.c: Call from 'some-username' (ip-of-sip.domain.com:5060) to extension 'some-username' rejected because extension not found in context 'my-context'.

The username that the provider gave me doesn’t exist in my own pbx’s extensions. I have inbound route to an existing extension. Perhaps, here I need to tweak SIP URI or something?

UPDATE: If I change the registration string to:
some-username@domain.com:some-password@@sip.domain.com:5060/my-local-active-extension

I am now able to receive incoming calls. The error however still appears in the logs.

8

That doesn’t make sense. If the error didn’t go away, one would expect it to change to say “to extension ‘my-local-active-extension’” and the call to fail. You can’t have a situation with that error message and the corresponding call being handled.

9

I know. It is funny thing. This provider is really strange to work with, but I am getting two DIDs for less than 3 USD/yearly so I can’t really complain. Their support doesn’t have much idea and denied giving me a proper trunk anyway.

The SIP URI they provide is some-username@domain.com , but unfortunately, as this is not a trunk, but just a SIP account I am unable to change it or tell it to ring something else on my asterisk box.

If I can’t figure it, I guess I will just let it pop an error on every call and leave it as-is.

With Groundwire, everything works fine - but I thought it would be nice to unify those two DIDs under one extension on my own PBX.

EDIT: You are a genius! I decided to restart Asterisk, just as try-out and the error went away! Now calls are being handled and no error in console anymore. Thanks, many many thanks! Your help was invaluable!

10

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