486 Busy Here

[ITSH]

I am attempting get freepbx to use my pots line connected to my obi110.  I have setup my obi110 following the standard directions (http://www.freepbx.org/support/documentation/howtos/howto-use-an-obi-110-device-to-provide-to-allow-freepbx-to-make-calls-o)  .  When I try to dial a number external the obi110 is sending a 486 back to freepbx.  Below is some error logging.  Any help would be greatly apperciated.


FREEPBX

[2015-05-16 21:59:17] VERBOSE[1913][C-00000001] netsock2.c: == Using SIP RTP TOS bits 184
[2015-05-16 21:59:17] VERBOSE[1913][C-00000001] netsock2.c: == Using SIP RTP CoS mark 5
[2015-05-16 21:59:17] VERBOSE[1913][C-00000001] app_dial.c: -- Called SIP/OBITRUNK1/17025551212
[2015-05-16 21:59:17] VERBOSE[1747][C-00000001] chan_sip.c: -- Got SIP response 486 "Busy Here" back from 192.168.123.215:5061
[2015-05-16 21:59:17] VERBOSE[1913][C-00000001] app_dial.c: -- SIP/OBITRUNK1-00000003 is busy
[2015-05-16 21:59:17] VERBOSE[1913][C-00000001] app_dial.c: == Everyone is busy/congested at this time (1:1/0/0)




OBI110 syslog
From: <sip:7025553434@192.168.123.210>;tag=as6636c051To: <sip:7025551212@192.168.123.215>;tag=SP2358cf20e6b7dccb4Via: SIP/2.0/UDP

192.168.123.210:5060;branch=z9hG4bK1dde1684;received=192.168.123.210;rport=5060Server: OBIHAI/OBi110-1.3.0.2872[May 16 21:52:10][192.168.123.215]<7> SIP

DLG reject: 486
[May 16 21:52:10][192.168.123.215]<7> sendto c0a87bd2:5060(375)
[May 16 21:52:10][192.168.123.215]SIP/2.0 486 Busy HereCall-ID: 152f6daf32b6dbe6071689c11beeb8b8@192.168.123.210:5060

[azrobert]

It looks like you are sending the call to SP2 on the OBi110.
What is your SP2 X_InboundCallRoute?
For a test change the SP2 X_InboundCallRoute to: ph
Does the OBi110 phone port ring?
If you don't need a phone attached to the OBi110, just look for flashing phone port LED.
If the phone port LED flashes, post the OBi110 Call History.

[ITSH]

I tried this before when trying to diagnose.  to make sure I was getting the same results, I changed LI to PH.  The green led lighted on the phone port, flashes and I get a ringing sound through freepbx.

erminal ID   SP2   PHONE1
Peer Name      
Peer Number   7025553434   
Direction   Inbound   Inbound
20:22:15   Ringing   
20:22:29   End Call   

[azrobert]

The call is getting to the OBi110.
The outbound number is also getting there correctly (peer number=7025553434).
If the InboundCallRoute is "li", the call should get routed to the POTS line.
I'm guessing there is a connection problem between the OBi110 and the POTS Line.
Do you have a phone you can connect to the OBi110 to confirm my theory?
Dialing # should get you dial tone from the POTS Line.

Are you located in North America?

[ITSH]

to go with my obi110 I wired a new phone jack.  to test the jack I was going by obi110 port status which said on hook.  shouldve tested it with a phone.  i was able to get a call to complete but it appears calls dont always go through. some are rendering just silence.  I remember seeing some articles talk about changing timers.  here are some interesting errors so far from freepbx

[2015-05-18 19:35:03] WARNING[1747] chan_sip.c: Timeout on x15ysFX9E90kA-NbFQimaw.. on non-critical invite transaction.
[2015-05-18 19:35:06] WARNING[1747] chan_sip.c: Retransmission timeout reached on transmission 7QcETL6EeT9R7hKaWodPIg.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2015-05-18 19:35:06] WARNING[1747] chan_sip.c: Hanging up call 7QcETL6EeT9R7hKaWodPIg.. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).