SIP phone B, on Asterisk server B can call to SIP Phone A on Asterisk Server A but not vice versa

As mentioned in the title, here I is the details I’ve stuck with this matter for 2-3 weeks now…

In my company there’s an Asterisk server setup previously (lets name it Server A) in HQ. My task was to setup 2 more asterisk server on our 2 remote company branches( lets name it Server B on Branch B, and server C on Branch C), so that the HQ staff can call the branches staff as if they’re in same building.

I have install the 2 servers B and C on branches B and C, I have configured everything, the SIP extension, iax2 trunks and works fine…meaning staff in branch B can communicate with staff in branch C, no delay or lagging what so ever great.

When I tried to connect ot HQ (server A) I have an issue, that is the extension in the branches can call staff in HQ, can talk and talk back…fine. HOWEVER the extension in HQ unable to call the extension on both branches B or C…keep getting “Your call cannot be completed as dial…”

I have stuck on this matter for quite some times now…have go through the config on the 3 servers, the firewall and even the company firewall…I even compare and tried various parameter change on the iax trunk setting many times…but still the same result.

Does anybody faced the same issue before?

Can anybody help me please?..I’m running out of area to look to troubleshoot.


Between A and [BC] what is? Public network, are A has public IP? Maybe firewall, what are CLI on A shows ?

There a lease line connection between A dan [BC] (IPVPN connection to be exact). I have tried disable the firewall but still no changes.

On CLI A show that the iax is established

From Server A

freepbx-b*CLI> iax2 show peers

Name/Username Host Mask Port Status Description

HQTOA/HQTOA (S) 4569 OK (8 ms)

HQTOB/HQTOB (S) 4569 OK (9 ms)

2 iax2 peers [2 online, 0 offline, 0 unmonitored]

Hope this provide clearer picture

Thank you in advance


So you have problem when you made call from [BC] to A? Are you check if IAX signalling are coming to server A?

As a technical point SIP severs cannot “connect” to anything, but I assume you meant Asterisk daemons running SIP user agents.

However more significant is your reference to IAX trunks. IAX is not SIP!

As you have provided neither configuration nor logs, this can only be a guess, but do you have type=friend and a local device with the same caller ID as one on he other machine?

Hi David,

This is the iax2 trunk config

On server A (HQ)

If this is chan_sip, use one section, with secret and remotesecret, and use type=peer. The need for separate incoming and outgoing sections is a misunderstanding propagated by cookbook configurations.

Actually, for a private trunk, there is no real point in having different secrets in each direction, in which case you can use one section with just secret. Many people would just rely on IP address checking.

Also for this to have worked in even one direction, you must have redacted relevant information, as a type=user section wouldn’t have worked at all unless the user part of the From header was “incoming”.