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Can't use trunks for outgoing calls every few days


Worm78
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I'm running 1.5.2.10 I keep running into an issue where I can't dial out on my sip or pots line trunks. Happens every 3-4 days. It happened twice right after I registered a user on a phone but I thought this was a coincidence. If I reset the pbxnsip service all is good again.

I have 9 set for my sip and 8 set for my gateway. If you try to dial it gives you a recording saying you are dialing an invalid extension. Trunks both show good and registered at this point. It normally happens mid day and I need to get us back up so I don't have much time for troubleshooting. Yesterday I upped the log levels and tried to make a call during the down time, then cleared the log and made another call after the service was reset. I don't see any errors jumping out however I'm failry new to voip. anyone see anything?

 

.0.0.0

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I found the issue. It appears all the co lines are staying active but only 1 person is on a call. We have 12 total lines. I had an issue with several co lines that did not disappear when deleting an old trunk so this may be a remaining issue.

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I found the issue. It appears all the co lines are staying active but only 1 person is on a call. We have 12 total lines. I had an issue with several co lines that did not disappear when deleting an old trunk so this may be a remaining issue.

 

Whow. Check the XML files in the colines directory - or just get rid of the CO-lines if you don't really need them. CO-lines were a pain in the neck in 1.5.

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Whow. Check the XML files in the colines directory - or just get rid of the CO-lines if you don't really need them. CO-lines were a pain in the neck in 1.5.

 

 

I checked all the xml files and removed the dups and redirections in the other folder with another forum assisted fix.

 

We removed all co lines then went in and removed the refs for the ramining ones listed in accounts. I then addedthe co lines back in.

 

So I can just delete all refrences to co lines in both trunks?

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I checked all the xml files and removed the dups and redirections in the other folder with another forum assisted fix.

 

We removed all co lines then went in and removed the refs for the ramining ones listed in accounts. I then addedthe co lines back in.

 

So I can just delete all refrences to co lines in both trunks?

 

I would first try to delete the CO lines in the trunk and see if there is anything left in the colines directory. If that should be the case, then just delete them in the directory and restart the service.

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I would first try to delete the CO lines in the trunk and see if there is anything left in the colines directory. If that should be the case, then just delete them in the directory and restart the service.

 

 

How would I setup my receptionist phone if i don't have co lines? If I remove them I get a busy signal once one call comes in. She uses a snom 360.

 

Scenario is trunks rolls to hunt group 700, rings 101 being the receptionist for 10 seconds then rolls to 701 which is the auto attendant, or 702 at night.

 

Without co lines, I called in and answered the call then took another cell and called in and I get a busy.

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Turn call waiting on the phone... Or use an agent group!

 

 

If using call waiting option will this allow for several calls to be put on hold? I was debating just call forwarding busy to the auto attendant. That way if she in on the phonethey don't let it ring for 10 seconds for no reason.

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I'm running 1.5.2.10 I keep running into an issue where I can't dial out on my sip or pots line trunks. Happens every 3-4 days. It happened twice right after I registered a user on a phone but I thought this was a coincidence. If I reset the pbxnsip service all is good again.

I have 9 set for my sip and 8 set for my gateway. If you try to dial it gives you a recording saying you are dialing an invalid extension. Trunks both show good and registered at this point. It normally happens mid day and I need to get us back up so I don't have much time for troubleshooting. Yesterday I upped the log levels and tried to make a call during the down time, then cleared the log and made another call after the service was reset. I don't see any errors jumping out however I'm failry new to voip. anyone see anything?

 

.0.0.0

 

You mean your cs410 is running 1.5.2.10, correct?

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