1

Remote Cisco SPA303 and a U100

Kealan Morrissey

March 23, 2017 12:26

None

Follow

Hello and thanks in advance for any help provided. I have a bit of an unusual one. 

 

We operate over two sites both running a u100 ( not paired ), one in each site.

Recently a genius decided instead of getting the standard handsets he would purchase some Cisco SPA303 's

I can get the SPA303 to register locally to its u100 on site no problem, but we route calls from the other site on line 2. So Line 2 needs to be a remote registration.

I have setup line 2 with all the appropriate details ( have a dozen other phones in this configuration that are not cisco, no problems there ). The phone registers and is visible on the remote u100 but the extension is unreachable when you try and dial or route calls to it. It just produces a single tone response and nothing further.   

I have followed the steps on this guide to the T after my own process failed - https://sysadminman.net/blog/2011/using-a-ciscolinksys-spa-504g-with-asterisk-and-freepbx-2714 

But all I get when I dial the extension is a single beep despite the remote u100 registering it as live, and the phone confirming that it is fully provisioned. 

 

Thanks in advance for any assistance here

• Facebook
• Twitter
• LinkedIn

3 comments

• 1
  
  Oscar Colka March 24, 2017 02:34

  A single beep sounds like the extension is setup as DND mode.

  Try register another extension and see what would be the result.

  If change the extension helps, then it is probably an issue of DND mode enable on the extension. Go to extension page of the U100 and disable the DND.

  If change the extension doesn't help, then it might not be the issue of DND. We will need further checking. If you can, try follow below steps and get some logs:

  1. Login via SSH to the U100

  See here for how to login SSH:

  https://support.yeastar.com/hc/en-us/articles/225066728-How-to-Login-the-SSH-of-Yeastar-MyPBX

  2. Enter asterisk CLI by excuting below command

  asterisk -vvvvvvvvvvvvvvr

  3. After you enter asterisk CLI, execute below commands:

  sip show peers

  4.  Make a call from remote site to the one which couldn't work;

  5. Copy the logs and send post them here

• 0
  
  Kealan Morrissey March 27, 2017 11:07

  Hi Oscar,

   

  I think your pretty much on the money with this it appears that the device is busy. I cant find any reason for this however. 

   

  [2017-03-27 11:06:54] NOTICE[3612]: app_mixmonitorcall.c:367 parse_mixmonitorrecord: fxo blf dst:

  == Spawn extension (queue-682, 682, 3) exited non-zero on 'SIP/trunk-GF051348461-0000109c'

  == End MixMonitorCall Recording /media/ysDisk_1/autorecords/20170327/20170327110436-Inbound-GF051348461-AAAB*-682(307)

  -- Got SIP response 486 "Busy Here" back from 46.10.238.156

  -- SIP/311-000010a3 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
  -- Executing [s@macro-realstexten:13] Goto("SIP/303-000010a2", "s-BUSY,1") in new stack
  -- Goto (macro-realstexten,s-BUSY,1)
  -- Executing [s-BUSY@macro-realstexten:1] GotoIf("SIP/303-000010a2", "1>0?follow-me-busy,1:1-out,1") in new stack
  -- Goto (macro-realstexten,follow-me-busy,1)
  -- Executing [follow-me-busy@macro-realstexten:1] GotoIf("SIP/303-000010a2", "1>0?vm-b,1") in new stack
  -- Goto (macro-realstexten,vm-b,1)
  -- Executing [vm-b@macro-realstexten:1] GotoIf("SIP/303-000010a2", "0?vm-b,2:vm-busy,1") in new stack
  -- Goto (macro-realstexten,vm-busy,1)
  -- Executing [vm-busy@macro-realstexten:1] VoiceMail("SIP/303-000010a2", "311,s") in new stack

  -- <SIP/303-000010a2> Playing 'beep.gsm' (language 'en_BR')

  -- Recording the message

  -- x=0, open writing: /var/spool/asterisk/voicemail/default/311/tmp/excwQ6 format: wav49, 0x570bf0

• 0
  
  Oscar Colka March 28, 2017 03:04

  Hello Kealan,

  Check if the extension is configured with any kind of DND or always folow me (forwarding), if you can please try capture a screenshot of the extension page, make sure the DND and always follow me setting is uncheck (means disable)