Asterisk SIP proxy in between one SIP and one PABX

[jenson]

Hello Everyone,
I’m very new and start exploring on Asterisk, please bear with me. I’m sure there is some of this topic in the forum but I would like to put in my scenario in again.I hope it does help me , again apologize and bear with me.

I’m trying to create an asterisk SIP proxy that sit in between a SIP server and my 3CX IP-PBX, all the IP-Phones are connected to my 3CX IP-PBX. SIP server on the other side are meant for some non IP-phone devices. There are some technical security concern why the IP-PBX can’t be connected directly to the SIP server. Issue here is I am not able to make call to go through via the Asterisk SIP Proxy that sit in the middle from both side.

My setup asterisk SIP proxy has 2 interfaces , one interface that can reach to SIP server and one interface that can reach my 3CX IP-PBX.

My sip.conf
[SIP]
type=friend
host=192.168.1.1 → my SIP IP
fromusername=pbx
fromdomain=SIP
username=pbx
secret=password
dtmfmode=rfc2833
allow=alaw
insecure=port,invite
context=incoming
directmedia=no

[3CX]
type=friend
host=10.240.7.100 → 3CX PABX
dtmfmode=rfc2833
allow=alaw
directmedia=no
insecure=port,invite
context=incoming

Do I need to configure extensions.conf since my dial plan all coming from 3CX IP-PBX?

Highly appreciate any advise.

[jcolp]

Asterisk is not a SIP proxy. It’s a B2BUA, and each call leg is independent. Yes, extensions.conf has to be configured as that defines how calls are handled.

You should also actually show or further describe what isn’t working/what happens.

[jenson]

Hi, thanks for the reply.

The problem in my described situation is, call from both side that is going through the Asterisk server that sit in between were not able to call up to each other.

How should I configure my extensions.conf?

[Pooh]

On Sunday 11 August 2024 at 15:05:57, jenson via Asterisk Community wrote:

Hello Everyone,
I’m very new and start exploring on Asterisk

My sip.conf

Since you are just starting with Asterisk, you really shouldn’t be using
sip.conf - it has been deprecated for many years, is no longer supported, and
doesn’t exist in the latest versions of Asterisk.

There is no point spending your time learning about this obsolete channel
driver; please use PJSIP from the start so that you are using something which
will continue to be supported, and about which you are likely to find more
people here with familiarity of it.

Antony.

–
“Can you keep a secret?”
“Well, I shouldn’t really tell you this, but… no.”

                                               Please reply to the list;
                                                     please *don't* CC me.

[jenson]

Hi Pooh,
Thanks for the tips. Unfortunately I have an old version of Asterisk that is left to me and I need to work on that. Still need to crack on the sip.conf and extensions.conf

[Pooh]

On Sunday 11 August 2024 at 15:57:15, jenson via Asterisk Community wrote:

Hi Pooh,
Thanks for the tips. Unfortunately I have an old version of
Asterisk

Which version?

that is left to me and I need to work on that.

Which distribution and version is it running on?

Still need to crack on the sip.conf and extensions.conf

PJSIP has been around for quirte some years, even if for some reason you need
to continue using an old (= also unsupported) version of Asterisk.

I’m not trying to be critical; it just sounds like a far from ideal starting
point for a beginner.

Antony.

–
Software development can be quick, high quality, or low cost.

The customer gets to pick any two out of three.

                                               Please reply to the list;
                                                     please *don't* CC me.