Something is limiting my number of concurrent calls (closed)

Hi there - something is limiting the number of concurrent calls out of my FreePBX 14.0.3.19 box. Symptom is “all circuits are busy” message when calling, and the provider (voip.ms) says the call in question never hits the service; the # of simultaneous calls never comes close to the provider limit.

FreePBX config is very vanilla - only the one SIP trunk to voip.ms. All config done through the UI. About the only thing I have changed is to use the “old” SIP signaling port 5060. Ports forwarded on our router, static IP address, FiOS fiber 25/25 Mbps service, all works - except can’t make over 5 simultaneous calls!

“Maximum Channels” under Trunks is blank.

Anything else to check? All thoughts welcome. Thanks!

Call trace of a failed call should reveal clues:

https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs

I think these are the pertinent lines from a call log. I can provide the full log if it helps (XXXX is my edit):

2018-10-16 12:22:27] VERBOSE[20710][C-00002703] app_dial.c: Called SIP/voipms/1516271_XXXX_
[2018-10-16 12:22:27] VERBOSE[20710][C-00002703] app_dial.c: SIP/voipms-0000575e is busy
[2018-10-16 12:22:27] VERBOSE[20710][C-00002703] app_dial.c: Everyone is busy/congested at this time (1:1/0/0)
[2018-10-16 12:22:27] VERBOSE[20710][C-00002703] pbx.c: Executing [[email protected]:26] NoOp("SIP/243-0000575d", "Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 17") in new stack

This says the outbound INVITE is reaching the provider and they rejected the call. Examining the SIP signalling might yield more info, but they should be able to tell you why the call was rejected.

OK - classic. voip.ms said “the call never reached them” - so they suspected a simultaneous call limit reached. sigh.

Is there a way for a “mortal” to check SIP signaling? I know how to use wireshark / pingplotter if that helps. Pointing at something to read would be fine.

If you’re on SNG7, sngrep will make it easy. If not already installed:

yum install sngrep

OK thank you! Let’s call this thread closed.

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Asterisk relates this to a 486 (user gave a busy) or 600 (user busy and there is no other option, like voicemail to go to). Both of those should be a result of the call hitting the destination (or at least their provider).

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