DISA calls not going outbound

Hi. We are on FreePBX 15.0.17.64. A single SIP provider, Vitelity.

Way back when, I setup a DISA so staff could call in and then turnaround and call outbound (as well as do other operations like voice mail). I’m talking 2009-ish, and it was used heavily for a few years.
Fast forward to a very large number of FreePBX updates and upgrades and now the outbound calls originated via DISA no longer work.

The key log entry where everything seems to go pear-shaped is here (for privacy reasons, I’m replacing the last four digits with xxxx)

– Executing [[email protected]:1] NoOp(“PJSIP/vitel-inbound-00000790”, “called 91520245xxxx in from-internal by ID: 1”) in new stack
– Executing [[email protected]:2] Dial(“PJSIP/vitel-inbound-00000790”, “Local/[email protected],300,T”) in new stack
– Called Local/[email protected]

If I make the same call (91520245xxxx) from a local extension, the log is simply:

-- Executing [[email protected]:1] Macro("PJSIP/105-0000078e", "user-callerid,LIMIT,EXTERNAL,") in new stack

The logs for both calls are very similar until the actual action happens, when the DISA call shows:

== Spawn extension (from-pstn, 91520245xxxx, 1) exited non-zero on ‘PJSIP/vitel-outbound-00000791’
– PJSIP/vitel-outbound-00000791 Internal Gosub(func-apply-sipheaders,s,1(9)) complete GOSUB_RETVAL=
– Called PJSIP/[email protected]
– No one is available to answer at this time (1:0/0/0)

(the “no one is available to answer” happens instantly; there is no call outbound to our SIP provider, Vitelity)

The local extension calling log looks identical at this point:

== Spawn extension (from-pstn, 91520245xxxx, 1) exited non-zero on ‘PJSIP/vitel-outbound-0000078f’
– PJSIP/vitel-outbound-0000078f Internal Gosub(func-apply-sipheaders,s,1(9)) complete GOSUB_RETVAL=
– Called PJSIP/[email protected]

I am throwing this out here because I can’t see any obvious thing, and my Google foo is not turning up anything with this funny “NoOp” return in the DISA call.

Does anyone have any pointers to where I should be looking?

Help much appreciated,

Joel

Please reply with a call trace via pastebin Providing Great Debug - Support Services - Documentation

Also, what version of FreePBX and Asterisk are you using?

This forum won’t let me put in a URl but… Here’s a trace (I think this is a full trace… let me know if I’m wrong) of the call that works: (from pastebin) fAHyrzhR
and here is the one from DISA that doesn’t work: (from pastebin) Lizh5vsN
Asterisk is “Asterisk 16.20.0” and FreePBX is freepbx.x86_64 14.1-1.sng7

Thanks for looking at it!!!
Joel

I am guessing that Vitelity is rejecting the call because you are attempting to send an unauthorized caller ID (the T-Mobile 520248xxxx number), presumably the number of the original caller. For the DISA in question, try setting Caller ID to your main number e.g.
"Opus One" <5203240494>
and retest.

If you had already done that or it doesn’t help, at the Asterisk command prompt type
pjsip set logger on
make another failing DISA call, paste the Asterisk log for the call (not the console output) at pastebin.freepbx.org and post the last eight hex characters of the link.

When you paste a log, we prefer you to use pastebin.freepbx.org, but whatever service you use, please take care that the expiration is set to Never, so future readers of this thread can follow along.

1 Like

Fantastic, yes, thanks Stewart, that was exactly the issue. CID not being overridden explicitly, combined with the “new” authorization going on. Thanks!

Thanks for the advice on pastebin; I’ll be sure to use the right service the next time around.

Appreciate the help and debugging!

Joel

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