Newbie - incoming call

Hi

I have tried to set up FreePBX on a Raspberry Pi2, but I am a newbiw on both Asterisk and FreePBX - and hope someone might guide me in the right direction - at least just so I can get a call through to a softphone.

I have configured a SIP trunk to an account at a local SIP provider (it is just a single PSTN number. probably mainly intended for connecting a softphone directly).

On incomming call routes, I have made a rule, that unconditionally should forward the call to extension 6002.

But when calling the PSTN number at the provider, the call don’t get through - and the Asterisk log says:

[Feb 12 11:47:05] WARNING[3055][C-00000044] chan_sip.c: username mismatch, have <trunk_1>, digest has <sip_provider_login_namej>
[Feb 12 11:47:05] NOTICE[3055][C-00000044] chan_sip.c: Failed to authenticate device “28282828” sip:[email protected];tag=as216d0c16

The 28282828 is the phonenumber of the cellphone used to call the PSTN number at the SIP provider.

The System Status shows my trunk as Registered, Asterisk is running 11.16.0

Setup is basically done from the guide at nerdvittles.com/?p=12233

Regards
Brian

Create an inbound route with the username of the account name VOIP:

Hi

So far, so good :smile:

I am now trying to get the outboud calling to work… My trunk has a match pattern of XXXXXXXX and no outbound prefix.
My outbound route (named Default-Out) has dial pattern XXXXXXXX and selects my trunk as first choice. Route CID is set to the PSTN number af the SIP provider, and “Override extension” has been selected.

When dialing a PSTN number form my extension 601, the log says:

[2015-02-14 13:05:04] VERBOSE[6005][C-0000000e] pbx.c: – Executing [[email protected]:22] Dial(“SIP/601-00000013”, “SIP/sip_username/22445566,300,”) in new stack
[2015-02-14 13:05:04] VERBOSE[6005][C-0000000e] netsock2.c: == Using SIP RTP TOS bits 184
[2015-02-14 13:05:04] VERBOSE[6005][C-0000000e] netsock2.c: == Using SIP RTP CoS mark 5
[2015-02-14 13:05:04] VERBOSE[6005][C-0000000e] app_dial.c: – Called SIP/sip_username/22445566
[2015-02-14 13:05:36] WARNING[2993] chan_sip.c: Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Request) – See https //wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response

My firewall should be SIP aware (it is a Cisco ASA) - so I will not assume, that it is causing the issue…

Any hints ?