Unable to dial out internationally

#141

I would expect something in the asterisk logs if calls are getting cutoff - do you see anything in the logs when a call drops?
Try calling just a number on the PBX that doesn’t go to the PSTN to see if the behaviour is identical.

(Honor's Haven It) #142

It shows the call hangs up as if I hit ‘end call’ when I clearly don’t. I’ve called my desk phone and it drops off, same with outbound dialing. this is weird…

(Greg Kujawa) #143

At this point it seems as if there are issues similar to peeling an onion. At least based on the info you’ve provided. If you can’t post log, debug, trace, etc. details for call examples then unfortunately this makes troubleshooting more of a guessing game.

If it was me, I would look into getting an IP-PBX expert to either come onsite or setup a remote session to get things ironed out once and for all. They can demonstrate what they are doing and how they are doing it. So that way you can “learn the ropes.”

Ultimately they can clean up not needing the 8 prefix for dialing out, being able to dial any internal, local, long distance, or international number desired, and likely help clean up any quirks of having multiple Adtrans for PSTN access.

(Honor's Haven It) #144

https://pastebin.freepbx.org/view/ff45c3f7

Here’s a pastebin of the asterisk logs when attempting to call from 3001 (The soft phone) to 3178 (my desk phone)

Line 271 appears to be where the line disconnects.

#145

In Asterisk SIP Settings, Local Networks is not correctly set to include the subnet of your PC.
For example, change it from 10.80.10.0/24 to 10.0.0.0/8
After Submit, you must also restart Asterisk. Note that the restart will drop any calls in progress, so do this while the system is idle.

If it still fails, at the Asterisk command prompt type
pjsip set logger on
make another failing call and paste the new log, which will now include a SIP trace.

(Honor's Haven It) #146

image

Finaly was able to get the system reset, the issue is persisting. It just loses track of the call, Logs to be posted next

(Honor's Haven It) #147

https://pastebin.freepbx.org/view/44587559

#148

Sorry, my apology; I confused your system with someone else’s. The pjsip logger command does not apply to your PBX.

At a root shell prompt, type
asterisk -rx 'sip set debug ip 10.10.10.97'
and you should see
SIP Debugging Enabled for IP: 10.10.10.97
(If your PC IP address changed, use the new value in the above command.)
Then, make another failing call and post the log.

(Honor's Haven It) #149

https://pastebin.freepbx.org/view/a44bb1b9

Here you go, enabled the setting, lets hope this shows you what’s going on.

#150

It would be less confusing if you removed the duplicate definition of the log file you are posting.

Are you still really using Elastix ?, it is long dead and buried. getting rid of any remaining Elastix dross, might well make things clearer.

(Honor's Haven It) #151

It is (unfortunately) the system we are stuck with. Trust me, If I could get a new system installed, I would. Sorry for how messy the logs are, they are a mess to sort through I know, but I appreciate you guys helping me out.

#152

You are not stuck with removing the duplicate definition though :wink:

(Honor's Haven It) #153

Alright, made another test call to cut out all the nonsense and grabbed it before anything else jump in the way

https://pastebin.freepbx.org/view/befe7799

#154

Two (redundant) entries for

  1. X-Asterisk-HangupCause: No user responding

(remove your duplicate definition of the full log

grep full /etc/asterisk/*log*

)

(Honor's Haven It) #155

Running the Grep command, I receive these

/etc/asterisk/chan_allogsm.conf:; fullname sets just the
/etc/asterisk/chan_allogsm.conf:; fullname: sets just the name part.
/etc/asterisk/chan_allogsm.conf:;fullname = My Name
/etc/asterisk/logger.conf.old_2014Jun26_230359:full => notice,warning,error,debug,verbose
/etc/asterisk/logger.conf.old_2014Oct05_175421:full => notice,warning,error,debug,verbose
/etc/asterisk/logger.conf.rpmnew:;full => notice,warning,error,debug,verbose,dtmf,fax
/etc/asterisk/logger_logfiles_additional.conf:full => debug,error,notice,verbose,warning
/etc/asterisk/logger_logfiles_custom.conf:full => notice,warning,error,debug,verbose

Which of these do I delete? I don’t want to brick anything here

#156

(it is grep NOT Grep linux is incredibly case sensitive, always bear this in mind as you move forward)

/etc/asterisk/logger_logfiles_custom.conf:full => notice,warning,error,debug,verbose

#157

Now figure out why 3001 ‘ain’t there’

Cause No. 18 - no user responding.
This cause is used when a called party does not respond to a call establishment message with either an alerting or connect indication within the prescribed period of time allocated.

What it means:
The equipment on the other end does not answer the call. Usually this is a misconfiguration on the equipment being called.

(Honor's Haven It) #158

I’ve called my desk phone & my cell with the soft phone, both of them end up dropping from the soft phones end. But when I call the soft phone, the call lasts for upwards of several minutes. I’ll try to post logs on a longer call, but this one irrelevant password bug for a different extension keeps rearing it’s ugly head every time I try to capture the logs.

#159

Did you fix 3100? (nothing is irrelevant just commonly undervalued)

(Honor's Haven It) #160

https://pastebin.freepbx.org/view/b2ea30d6

Here’s the office phone calling the soft phone. As for the password error, the extension flat out doesn’t exist in the main system, so I’ll have to add them eventually. For now, lets focus on the task at hand.