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.
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.
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.
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.
/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
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.
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.
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.
Yes, look at line 11: Contact: <sip:[email protected]:5060>
Asterisk is supplying a public IP address, because it does not recognize that 10.10.10.97 is local, even though you presumably updated that setting and restarted Asterisk.
In the file /etc/asterisk/sip_general_additional.conf please post the value of any lines beginning with localnet
Also, in the Elastix GUI, post a screenshot of Settings → Advanced SIP Settings → NAT.
