QUESTION 34 - asterisk everyone is busy/congested at this time

PBXEHR

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In this instance VOIP works, but inbound and outbound POTS stopped working.

I can run reload at the Asterisk prompt and everything is fine again.

This happens periodically, trying to figure out why.
 

Support MTC

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Sometimes this occurs with running background tasks and sometimes this occurs due to instability of the system ( such as hardware malfunction). PLease make sure you do not have a compromised system first and all patches and updates have been applied. After a clean reboot you should see when the problem occurs again and pull the full log file to see if it is an asterisk issue or a system problem. hope this helps
 

PBXEHR

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It is patched and updated every day. Has explicit White List of allowed IP's that may connect to it. Not compromised. Software is running on Centos 6.5 as a Microsoft Hyper-V Virtual Machine. VOIP consistently works, it's the POTS connected via Linksys 3102's that's seems to be flakey.
 

MGD4me

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The 3102 should be configured as a Trunk to the PBX. Does 'Asterisk Info' show that the PSTN (FXO) trunk is registered?
 

PBXEHR

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I can't find an Asterisk Info command. Here's what Chan Sip Peers shows and the POTS lines are currently down again.
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
196/196 10.254.254.71 D No No A 5067 OK (10 ms)
197/197 10.254.254.73 D No No A 5065 OK (11 ms)
198/198 10.254.254.70 D No No A 5063 OK (6 ms)
199/199 10.254.254.72 D No No A 5061 OK (12 ms)

1234567/asterisk 10.254.254.72 Yes Yes 5061 Unmonitored
8061234568/asterisk 10.254.254.73 Yes Yes 5065 Unmonitored
8061234569/asterisk 10.254.254.70 Yes Yes 5063 Unmonitored
8061234560/asterisk 10.254.254.71 Yes Yes 5067 Unmonitored
(I changed the actual phone #'s)

Active Channels
Channel Location State Application(Data)
SIP/504-00000468 s@macro-dialout-trun Up Dial(SIP/1234567/1877123134,3
SIP/1234567-00000469 (None) Up AppDial((Outgoing Line))
2 active channels
1 active call

Chan Sip Channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
10.254.254.63 504 ab22eead-c211b8 (ulaw) No Rx: ACK 504
10.254.254.73 (None) ab510123-432fa9 (nothing) No Rx: REGISTER <guest>
10.254.254.72 (None) dec5e234-1232aa (nothing) No Rx: REGISTER <guest>
10.254.254.72 18779326301 123e8d514432c11 (ulaw) No Tx: ACK 7981212
4 active SIP dialogs

Ok here's what happened now.
When I dial out on POTS, I get the all circuits are busy now and someone was on the main line. They hung up from their call, when I dialed again, the call went through.
So it's like out outgoing call isn't rolling over to the next line when the 1st POTS line is already on a call.
 

MGD4me

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So it's like out outgoing call isn't rolling over to the next line when the 1st POTS line is already on a call.

Okay, the picture is getting a little less hazy as we get more information.

You have four 3102 ATAs, all configured as individual Trunks, then? I would have to believe that you also have four individual POTS lines as well, one separate line for each 3102?

Let's have a look at your Outgoing Route, then. You *should be able to* (tm) plug all four 3102 Trunks into the Outgoing Route, in any particular order to suit your needs. Let's start there. And perhaps you may want to scroll through the logs as a call is being placed.
 

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