Several extensions showing UNKNOWN status in CHAN_SIP PEERS
On Grandstream each ATA must have it's own SIP port if they are on the same LAN. 5060, 5061, 5062, etc... You can also try the setting "use random SIP port".
View ArticleTwo addresses for NAT
Hi, You can try to use the Resiprocate proxy. Thank you, Daniel FriedmanTrixton LTD.
View ArticleTwo sip trunks same provider different area codes
Does anyone know what and/or how to declare the tgrp options for trunk groups in freepbx/asterisk? It is a GenBand C15 that is providing the SIP Trunk. We are troubleshooting one of two options. A)...
View ArticleWrong Server Time
SSH into the server. Check the status of the ntpd serviceservice ntpd status Is ntpd running ? if not start the service and ntpd will start checking in for time updates. If you want to update time now...
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Upgrading to stable 13, puzzled
I ran the updates to 6 offered by System Admin. Then I ran the update under ADMIN menu and now Sys Admin shows 6.12.65-31 and no further updates are listed. I have looked at related entries in the...
View ArticleReload failed because retrieve_conf encountered an error: -1
Yes it is, didn't have it installed, sorry for being a idiot just getting started with all this.
View ArticleReload failed because retrieve_conf encountered an error: -1
No problem... Have a nice day! Nick
View ArticleUpgrading to stable 13, puzzled
Hi! Maybe there is a way to change track using System Admin Pro and if so I don't know it but what I would do next is run https://upgrades.freepbxdistro.org/stable/10.13.66/upgrade-10.13.66-1.sh see...
View ArticleUpgrading to stable 13, puzzled
Correct for sanity reasons we make you run the upgrade script to the next track manually. Then sysadmin can keep it updated after that for the current 10.13.66 track
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No audio on both inbound or outbound calls
Hello I've installed 12.0.76.2 and Asterisk 13 on my centos 6.7 machine, and everything seemed to be in order as I followed the official install guide, with some small issues here and there. This is...
View ArticleIAX2 Extensions and auto answer header
I am using an IAX2 extension on a Zoiper softclient. I would like to send an auto answer header to my extension. Can I send auto answer headers in IAX2 at all?
View ArticleHelp Understanding Version Numbers
So i recently upgraded to version 13 using the upgrade module in the web gui. I'm still on Asterisk 11 which is causing some issues with CDR's and User Manager. I was going to try and upgrade my...
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Help Understanding Version Numbers
Hi! munozj: 1) what is the the actual latest stable version for me to be running and 10.13.66-6 for the distro... As for the FreePBX modules themselves they each have their version... munozj: 2) how...
View ArticleError upgrading ucpnode and voicemail modules on v.13
I'm using FreePBX 13 and for some reason I'm trying to upgrade the ucpnode and voicemail modules and get a similar error. Downloading and Installing ucpnodeDownloading ucpnode Error(s) downloading...
View ArticleSorting Voicemails Newest to Oldest
Currently on the Voicemail Rest App, voicemails are sorted oldest at the top... is there a way to change this to be newest at the top? This is on an Aastra/Mitel 6867i
View ArticleError upgrading ucpnode and voicemail modules on v.13
Hi! Looks like it is trying to to is a "rm -rf" and it fails.. I would suggest permission problems? I would probably issue a "fwconsole chown" and retry... By the way, are we talking of the distro...
View ArticleError upgrading ucpnode and voicemail modules on v.13
I think that worked.Ran it and it removed a bunch of "dangling symlink"s and module no longer needs to be upgraded. Thanks, learned something new!
View ArticleError upgrading ucpnode and voicemail modules on v.13
You are welcome! Have a nice day! Nick
View ArticleNo audio on both inbound or outbound calls
For anyone having the same issue here is the solution to get back sound if you're calling the same server extension:...
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No audio on both inbound or outbound calls
Final solution that worked for me is that I enabled IP to make an outbound call. I use a flowroute so you have to do that explicitly in order for your sound to go trough. You can call but there is no...
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