Credential Failure Connecting Linphone to FreePBX

EdArmstrong · March 6, 2019, 8:24pm

Hi - hoping someone can help me with this one, I’ve started work on a project to turn a Raspberry Pi into an intercom system for my flat. The current system works through our telephone line so I thought it might be efficient to setup a VoIP line for this purpose rather than run the intercom through a paid land line, that we never use for other purposes.

As you can probably tell I’m pretty new to all of this so have been using the “voip-for-dummies” guide on the littlebytesofpi website (i’m a new user so can’t post links) to setup the VoIP server etc, however the guide is a little outdated now and so the steps don’t appear to be as easy to follow as I presume they once were.

I’ve gone through the steps to install Asterisk (v13.20.0) and FreePBX (v14.0.2.10) onto my Pi, which all seems to have worked as expected. I’ve also run the suggested updates and have setup a couple of extensions when accessing the server via a browser: raspbx.local

I now want to test connectivity to the server but I’m having no luck getting my SIP softphone (i’m using the Linphone app on my Android mobile) to register with the server.

The guide I used suggests that I need to enter a username, password & domain combination but every time I do so I am met with a “Registration failed” error message.

I have tried many, many different combinations but have yet to hit the correct combination in order to register successfully.

Here are some of the variables I am working with:

The guide suggests I simply select “SIP” as the extension protocol however, FreePBX appears to offer two types of SIP: Chan_SIP and PJSIP - I understand PJSIP is a newer version of the protocol however I’m not sure which I should be selecting in the case of this setup (or if it matters at all).
Username - when I setup a new extension via the raspbx portal I am asked to provide a " User Extension " as well as a " Display Name " - which of these (if either) is the username? To simplify things for my test I simply named both entries as " 11111 ".
Password - there seem to be a few options of which password could be used here. It could be the password I set to log onto the server or it could be what is listed in the extension setup as " Secret " or " Password For New User " - again, I made sure all three were identical for my test to reduce the amount of combinations I needed to check to connect but with no success.
Domain - there seem to be even more options I could go for in this section, here’s some of the ones I’ve tested so far:

raspbx.local (adding spaces because I am not allowed to post links)
root @ raspbx.local
the inet ip addess under the “wlan0” section I see when I run ifconfig on my Pi
the net ip addess under the “lo” section I see when I run ifconfig on my Pi (localhost IP)
the ip address I see when I log onto the server via a web browser, listed as: **FreePBX {ip address} ‘VoIP Server’
11111 @ raspbx.local
11111@{all of the other IP addresses mentioned above}

Unfortunately nothing is working so I’m clearly doing something very wrong here. Can anyone advise what I should be trying in order to successfully connect the softphone to the server?

Let me know if any further detail is required.

Many thanks,

Ed

Stewart1 · March 7, 2019, 3:17am

Please try these settings in Linphone:

Username: match what you entered for User Extension in FreePBX (presumably 1111).
Password: match what you set for Secret in FreePBX. For initial testing, use a short string, for example zzxg2718 so you can be sure it’s correct.
Domain: match the IPv4 address you see for wlan0, for example 192.168.1.234

This is for a pjsip extension. If you created a chan_sip extension, also add:
4. Proxy: your wlan0 address followed by :5160, for example 192.168.1.234:5160

If you still have trouble, post what if anything appears in the Asterisk log when you attempt to register.

EdArmstrong · March 10, 2019, 12:36pm

Hi Stewart,

Thanks very much for this. I tried both of the ways you have described above (using pjsip and chan_sip) but unfortunately neither worked.

I have pasted the log for the attempted pjsip connection below - not sure why I can’t just upload a text file here but the forum only allows the upload of picture files, therefore apologies in advance for the particularly long post!

I’m guessing this is a big part of the issue:

ERROR[4651] res_pjsip_endpoint_identifier_ip.c: Identify ‘11111-identify’ is not configured to match anything.

Though I have definitely created an extension named ‘11111’ as you can see in the attached image

Any further advise would be much appreciated!

Thanks

Ed

Full Log:

[2019-03-10 10:53:37] VERBOSE[4215] loader.c: Reloading module ‘chan_iax2.so’ (Inter Asterisk eXchange (Ver 2))

[2019-03-10 10:53:37] NOTICE[4215] iax2/provision.c: No IAX provisioning configuration found, IAX provisioning disabled.

[2019-03-10 10:53:37] VERBOSE[4215] loader.c: Reloading module ‘chan_motif.so’ (Motif Jingle Channel Driver)

[2019-03-10 10:53:37] VERBOSE[4215] loader.c: Reloading module ‘chan_sip.so’ (Session Initiation Protocol (SIP))

[2019-03-10 10:53:37] VERBOSE[1187] chan_sip.c: Reloading SIP

[2019-03-10 10:53:37] VERBOSE[4215] loader.c: Reloading module ‘res_pjsip_outbound_registration.so’ (PJSIP Outbound Registration Su$

[2019-03-10 10:53:37] ERROR[4215] res_pjsip_config_wizard.c: Unable to load config file ‘pjsip_wizard.conf’

[2019-03-10 10:53:37] VERBOSE[4215] loader.c: Reloading module ‘res_pjsip_notify.so’ (CLI/AMI PJSIP NOTIFY Support)

[2019-03-10 10:53:37] VERBOSE[4215] loader.c: Reloading module ‘res_adsi.so’ (ADSI Resource)

[2019-03-10 10:53:37] VERBOSE[4215] loader.c: Reloading module ‘app_confbridge.so’ (Conference Bridge Application)

[2019-03-10 10:53:37] NOTICE[4215] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge

[2019-03-10 10:53:37] VERBOSE[4215] loader.c: Reloading module ‘cel_manager.so’ (Asterisk Manager Interface CEL Backend)

[2019-03-10 10:53:37] VERBOSE[1187] netsock2.c: Using SIP TOS bits 96

[2019-03-10 10:53:37] VERBOSE[4215] loader.c: Reloading module ‘cel_odbc.so’ (ODBC CEL backend)

[2019-03-10 10:53:37] VERBOSE[1187] netsock2.c: Using SIP CoS mark 4

[2019-03-10 10:53:37] VERBOSE[4215] cel_odbc.c: Found CEL table cel@asteriskcdrdb.

[2019-03-10 10:53:37] VERBOSE[4215] loader.c: Reloading module ‘codec_speex.so’ (Speex Coder/Decoder)

[2019-03-10 10:53:37] VERBOSE[4215] loader.c: Reloading module ‘app_voicemail.so’ (Comedian Mail (Voicemail System))

[2019-03-10 10:53:37] WARNING[4215] app_voicemail.c: maxsilence should be less than minsecs or you may get empty messages

[2019-03-10 10:53:37] VERBOSE[4215] loader.c: Reloading module ‘app_playback.so’ (Sound File Playback Application)

[2019-03-10 10:53:37] VERBOSE[4215] loader.c: Reloading module ‘app_queue.so’ (True Call Queueing)

[2019-03-10 10:53:37] NOTICE[4215] app_queue.c: No queuerules.conf file found, queues will not follow penalty rules

[2019-03-10 10:53:37] VERBOSE[4215] asterisk.c: Remote UNIX connection disconnected