SOLVED Unable to login from ext 701 New install

[[email protected]]

I have followed the instructions from the Nerd Vittles site to install a new install on Digital Ocean.
All went well. I can log into web session. I cannot get a ext to register using the Yate Cleint.
I get timeout message. I have deployed this many times but cannot get my ext to login.

Any ideas?

I have any older Digital Ocean image that has been working or over a year, but besides the two extensions that are working on my Grandstream 2200 I cannot get any additional extensions to log on either, with phone or Yate client.

Something is stopping new ext to register. Tried every thing I can think of.

Thanks for any help!

 

[wardmundy]

What does the Asterisk CLI show when you attempt to register the Yate Client? Have you added the IP address of your Yate Client to the firewall whitelist using /root/add-ip?

 

[DoctorJ]

Having same issue. Was working on Wednesday evening. About 1am PDT on Thursday, all my phones lost registration. Can't get them to log back. Have my home IP in iptables. Can SSH into server.

There's an additional IP address on the status screen. A local IP --> 10.12.0.5

Could this be the problem?

 

[DoctorJ]

What I've noticed:
I can initially register an extension. However, after adding a Google Voice trunk, CHAN_PJSIP is disabled. In the extension dialog, I see:

CRITICAL ERROR! Required Service CHAN_PJSIP is disabled! This device is unusable!

After this, the originally added softphone will remain connected, but new extensions cannot connect. Also, if I disconnect the previously working softphone, it will no longer reconnect. I've gone to advanced settings, and re-enabled CHAN_PJSIP, but no joy. Error message persists...

Thoughts?

-----
UPDATE: Managed to get to work briefly by changing service to CHAN_PJSIP exclusively in advanced settings. Lasted maybe an hour. Attempted a few incoming calls. Then failed again with same message listed above...

 

[[email protected]]

I tried /root/add-ip. Still no luck. My original DO deployment only has two working extensions. When I try to add additional extensions it logs on for a short period of time only if I set the transport to TCP, then says unauthorized. Same on the new system I just set up.
Here is the debug messages on Yate. If I use UDP transport all I get is a timeout.

REGISTER sip:104.131.42.xx SIP/2.0

Contact: <sip:[email protected]:57728>

Expires: 600

To: <sip:[email protected]>

Call-ID: [email protected]

Via: SIP/2.0/TCP 192.168.1.110:57728;alias;rport;branch=z9hG4bK517745359

From: <sip:[email protected]>;tag=1994397201

CSeq: 591 REGISTER

User-Agent: YATE/4.0.1

Max-Forwards: 70

Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO

Content-Length: 0



------

<sip:INFO> 'tcp:192.168.1.110:57728-104.131.42.97:5060' received 326 bytes SIP message [0xe2776e0]

------

SIP/2.0 403 Forbidden

Via: SIP/2.0/TCP 192.168.1.110:57728;rport=57728;received=98.31.1.xxx;branch=z9hG4bK517745359;alias

Call-ID: [email protected]

From: <sip:[email protected]>;tag=1994397201

To: <sip:[email protected]>;tag=z9hG4bK517745359

CSeq: 591 REGISTER

Server: FPBX-12.0.70(13.7.2)

Content-Length: 0



------

<sip:INFO> Detected local address 98.31.1.xxx:57728 for SIP line 'sip:[email protected]'

<sip:WARN> SIP line 'sip:[email protected]' logon failure 403: Forbidden

 

[[email protected]]

Here are the Asterisk CLI messages:



=========================================================================

Connected to Asterisk 11.15.0 currently running on piafUbuntu (pid = 1777)

[2016-03-27 19:23:49] ERROR[17765]: chan_sip.c:16911 register_verify: 'TCP' is not a valid transport for '202'. we only use 'UDP'! ending call.

[2016-03-27 19:23:49] NOTICE[17765]: chan_sip.c:28104 handle_request_register: Registration from '<sip:[email protected]>' failed for '98.31.1.215:55088' - Device not configured to use this transport type

[2016-03-27 19:23:51] ERROR[17482]: tcptls.c:875 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.1.176:55737: Connection timed out

[2016-03-27 19:24:50] ERROR[17808]: chan_sip.c:16911 register_verify: 'TCP' is not a valid transport for '202'. we only use 'UDP'! ending call.

[2016-03-27 19:24:50] NOTICE[17808]: chan_sip.c:28104 handle_request_register: Registration from '<sip:[email protected]>' failed for '98.31.1.215:55108' - Device not configured to use this transport type

[2016-03-27 19:25:30] ERROR[17726]: tcptls.c:875 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.1.176:55738: Connection timed out

piafUbuntu*CLI>

 

[qtlnx]

Not sure if related but on one of several fresh installs from IncrediblePBX13.2.iso I went into user manager, user 701 and found the same
CRITICAL ERROR! Required Service CHAN_PJSIP is disabled! This device is unusable!

 

[hecatae]

Not sure if related but on one of several fresh installs from IncrediblePBX13.2.iso I went into user manager, user 701 and found the same
CRITICAL ERROR! Required Service CHAN_PJSIP is disabled! This device is unusable!

using ubuntu?

 

[qtlnx]

using ubuntu?

no, IncrediblePBX13.2.iso installs RHEL 6.x clone

 

[qtlnx]

Someone just closed http://pbxinaflash.com/community/threads/unable-to-login-from-ext-701-new-install.18957/ before I was able to report that upon re-running the same IncrediblePBX13.2.iso full install on the same system extension 701 looks normal:
This device uses PJSIP technology listening on udp:0.0.0.0:5060
and no errors/warnings on the screen.