Make sure that you name the trunk with numbers only and use that name as the autentication name on the gateway.

Hi Community,

I recently run an upgrade from 2.11 to v12 in the hope that we can slowly moved towards a distro of v14.
As after reading a few forums I see commercial modules are no longer supported on non distro versions.
I now have a problem I would like to sort out before moving to the distro after using the upgrade module.
EPM v12.x.x.x does not display correctly. Basefile management is blank, template line key options are missing at the bottom of the template page with only extension modules options listed. Also the extension mapping page looks abnormal. I have updated all the modules but could also be missing a module that EPM might require. I would appreciate some help if possible as I’m not sure if this could also be a php issue or similar.

Asterisk Ver: 11.17.1
Freebpx: 12.0.76.4
Centos: 6.9
PHP: 5.3.3

extension_mapping.jpg1607x406 116 KB

12 is not supported and has not been for years. You need to get together to 14.

Script to migrate settings from unsupported versions:

https://wiki.freepbx.org/display/PPS/Elastix+and+PBXinaFlash+to+FreePBX+Distro+Conversion+Tool

Thanks for the quick reply guys.
Should I attempt to migrate from v12 or roll back to 2.11then migrate. I can remove a snapshot and go back to a system that was running sweet before hand if need be.

Upon further review of Polycom’s awkwardly-organized documentation (including the first time I’ve seen configuration entries broken up into multiple lines making it very difficult to search for them in the PDF - there’s a technical writer out there who hates sysadmins) it looks like expansion module entries must reference directory entries, and that rabbit hole is pretty intense in and of itself.

If I get a working config built I’ll engage with the developers to get this sorted.

I am trying to setup a pjsip extension on my home office test system. The system seems to be up to date. When i run “cat /etc/schmooze/pbx-version” i get “10.13.66-22” back from the system.

The issue is when I build the extension using the Extensions Module within freepbx I cannot get any phone to register with that extension. I’ve tried a grand steam phone and a couple of soft phones.

All I keep getting is a 401 error and the asterisk terminal says “Request ‘REGISTER’ from ‘<1104…>’ failed for ‘IP’ (callid: qDKBOXaNp1sheYCteGbUtA…) - Failed to authenticate” and [2018-05-13 22:55:15] NOTICE[22363]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘REGISTER’ from ‘<1104…>’ failed for ‘IP’ (callid: e1DbMLQdM6HdnWCUtZfWdQ…) - No matching endpoint found

So after countless hours of scratching my head, and looking for answers, i decided to manually configure build a pjsip extension using the pjsip_custom.conf files in config edit. This worked and i managed to register the extension.

Next i copied the config freepbx generates when i build the extension within the extensions module and copied it into the custom config area, then deleted the ext from extensions module to not cause any problems. Then i troubleshooted the config and i came to the conclusion that the extension module is generating some code that asterisk does not like. On my system it creates a line “auth=1104-auth” and then a “[1104-auth]” entry with the username and password.

When i manually build the same exact config as the extensions module the registration fails in the same manner, however if i tweak “auth=1104-auth” to be “auth=auth1104” and “[1104-auth]” to be “[auth1104]” the extension will register correctly. However now i can’t use all the options that freepbx has to offer and now i have to configure dial plan manually.

Is this something anyone else has seen? Is my system running an outdated module? Any insight or advice would be appreciated.

Here is a list of the modules and their version numbers on my system

±---------------------±------------±----------------------------------±-----------+
| Module | Version | Status | License |
±---------------------±------------±----------------------------------±-----------+
| accountcodepreserve | 13.0.2.2 | Enabled | GPLv2 |
| announcement | 13.0.7.1 | Enabled | GPLv3+ |
| areminder | 13.0.10.6 | Enabled | Commercial |
| arimanager | 13.0.4 | Enabled | GPLv3+ |
| asterisk-cli | 13.0.4 | Enabled | GPLv3+ |
| asteriskinfo | 13.0.7.1 | Enabled | GPLv3+ |
| backup | 13.0.27.18 | Enabled | GPLv3+ |
| blacklist | 13.0.14.8 | Enabled | GPLv3+ |
| bria | 13.0.20 | Enabled | Commercial |
| broadcast | 13.0.12.10 | Enabled | Commercial |
| builtin | | Enabled | |
| bulkdids | 13.0.2 | Enabled | GPLv3+ |
| bulkextensions | 13.0.3 | Enabled | GPLv3+ |
| bulkhandler | 13.0.14.4 | Enabled | GPLv3+ |
| callback | 13.0.5.2 | Enabled | GPLv3+ |
| callerid | 13.0.8.9 | Enabled | Commercial |
| callforward | 13.0.4.2 | Enabled | AGPLv3+ |
| calllimit | 13.0.5.5 | Enabled | Commercial |
| callrecording | 13.0.11.4 | Enabled | AGPLv3+ |
| callwaiting | 13.0.4.1 | Enabled | GPLv3+ |
| campon | 13.0.4.1 | Enabled | GPLv3+ |
| cdr | 13.0.32 | Enabled | GPLv3+ |
| cel | 13.0.26.2 | Enabled | GPLv3+ |
| certman | 13.0.37.7 | Enabled | AGPLv3+ |
| cidlookup | 13.0.12.2 | Enabled | GPLv3+ |
| conferences | 13.0.23.11 | Enabled | GPLv3+ |
| conferencespro | 13.0.27.7 | Enabled | Commercial |
| configedit | 13.0.7.1 | Enabled | AGPLv3+ |
| contactmanager | 13.0.42.12 | Enabled | GPLv3+ |
| core | 13.0.122.16 | Enabled | GPLv3+ |
| cos | 13.0.12.1 | Enabled | Commercial |
| customappsreg | 13.0.5.4 | Enabled | GPLv3+ |
| cxpanel | 13.0.5.1 | Enabled | GPLv3 |
| dahdiconfig | 13.0.33.13 | Enabled | GPLv3+ |
| dashboard | 13.0.25.3 | Enabled | AGPLv3+ |
| daynight | 13.0.15 | Enabled | GPLv3+ |
| dictate | 13.0.5 | Enabled | GPLv3+ |
| digium_phones | | Not Installed (Locally available) | GPLv2 |
| digiumaddoninstaller | | Not Installed (Locally available) | GPLv2 |
| directory | 13.0.19.5 | Enabled | GPLv3+ |
| disa | 13.0.6.1 | Enabled | AGPLv3+ |
| donotdisturb | 13.0.3.1 | Enabled | GPLv3+ |
| endpoint | 13.0.118.21 | Enabled | Commercial |
| extensionroutes | 13.0.10.5 | Enabled | Commercial |
| extensionsettings | 13.0.4 | Enabled | GPLv3+ |
| fax | 13.0.40.5 | Enabled | GPLv3+ |
| faxpro | 13.0.38.9 | Enabled | Commercial |
| featurecodeadmin | 13.0.6.4 | Enabled | GPLv3+ |
| findmefollow | 13.0.38.11 | Enabled | GPLv3+ |
| firewall | 13.0.55.1 | Enabled | AGPLv3+ |
| framework | 13.0.195 | Enabled | GPLv2+ |
| freepbx_ha | 13.0.11 | Enabled | Commercial |
| fw_langpacks | 12.0.7 | Enabled | GPLv3+ |
| hotelwakeup | 13.0.17.1 | Enabled | GPLv2 |
| iaxsettings | 13.0.6.6 | Enabled | AGPLv3 |
| infoservices | 13.0.1.2 | Enabled | GPLv2+ |
| irc | 2.11.0.7 | Enabled | GPLv3+ |
| ivr | 13.0.27.7 | Enabled | GPLv3+ |
| languages | 13.0.6 | Enabled | GPLv3+ |
| logfiles | 13.0.10.4 | Enabled | GPLv3+ |
| manager | 13.0.2.5 | Enabled | GPLv2+ |
| miscapps | 13.0.3.1 | Enabled | GPLv3+ |
| miscdests | 13.0.5 | Enabled | GPLv3+ |
| motif | 13.0.3.2 | Enabled | GPLv3+ |
| music | 13.0.22.3 | Enabled | GPLv3+ |
| outroutemsg | 13.0.2.1 | Enabled | GPLv3+ |
| paging | 13.0.26.5 | Enabled | GPLv3+ |
| pagingpro | 13.0.19.7 | Enabled | Commercial |
| parking | 13.0.19.8 | Enabled | GPLv3+ |
| parkpro | 13.0.30.13 | Enabled | Commercial |
| pbdirectory | 2.11.0.6 | Enabled | GPLv3+ |
| phonebook | 13.0.6.1 | Enabled | GPLv3+ |
| phpinfo | 13.0.2 | Enabled | GPLv2+ |
| pinsets | 13.0.8 | Enabled | GPLv3+ |
| pinsetspro | 13.0.9.8 | Enabled | Commercial |
| pm2 | 13.0.5 | Enabled | AGPLv3+ |
| pms | 13.0.2.14 | Enabled | Commercial |
| presencestate | 13.0.8 | Enabled | GPLv3+ |
| printextensions | 13.0.3.1 | Enabled | GPLv3+ |
| queueprio | 13.0.2 | Enabled | GPLv3+ |
| queues | 13.0.34.9 | Enabled | GPLv2+ |
| qxact_reports | 13.0.15.6 | Enabled | Commercial |
| recording_report | 13.0.24.6 | Enabled | Commercial |
| recordings | 13.0.30.12 | Enabled | GPLv3+ |
| restapi | 13.0.21.1 | Enabled | AGPLv3 |
| restapps | 13.0.92.5 | Enabled | Commercial |
| ringgroups | 13.0.23.2 | Enabled | GPLv3+ |
| rmsadmin | 13.0.14.1 | Enabled | Commercial |
| sangomacrm | 13.0.4.30 | Enabled | Commercial |
| setcid | 13.0.6.2 | Enabled | GPLv3+ |
| sipsettings | 13.0.27.1 | Enabled | AGPLv3+ |
| sipstation | 13.0.14.8 | Enabled | Commercial |
| sms | 13.0.12.3 | Enabled | Commercial |
| sng_mcu | 13.0.5 | Enabled | Commercial |
| soundlang | 13.0.24.5 | Enabled | GPLv3+ |
| speeddial | 2.11.0.4 | Enabled | GPLv3+ |
| superfecta | 13.0.4.5 | Enabled | GPLv2+ |
| sysadmin | 13.0.76.4 | Enabled | Commercial |
| timeconditions | 13.0.34.9 | Enabled | GPLv3+ |
| tts | 13.0.10 | Enabled | GPLv3+ |
| ttsengines | 13.0.7.3 | Enabled | AGPLv3 |
| ucp | 13.0.42.5 | Enabled | AGPLv3+ |
| ucpnode | 13.0.34.9 | Enabled | Commercial |
| userman | 13.0.76.37 | Enabled | AGPLv3+ |
| versionupgrade | 13.0.1.2 | Enabled | Commercial |
| vmblast | 13.0.8 | Enabled | GPLv3+ |
| vmnotify | 13.0.22 | Enabled | Commercial |
| voicemail | 13.0.54.20 | Enabled | GPLv3+ |
| voicemail_report | 13.0.13.3 | Enabled | Commercial |
| vqplus | 13.0.39 | Enabled | Commercial |
| weakpasswords | 13.0.2 | Enabled | GPLv3+ |
| webcallback | 13.0.11.2 | Enabled | Commercial |
| webrtc | 13.0.32.8 | Enabled | GPLv3+ |
| xmpp | 13.0.17.13 | Enabled | AGPLv3 |
| zulu | 13.0.53.4 | Enabled | Commercial |
±---------------------±------------±----------------------------------±-----------+

Hi,

I’m running the following FreePBX Distro:
Your Linux Distribution: (Redhat Sangoma Linux release 7.4.1712 (Core) )
Your FreePBX version: (14.0.3.1)

All the inbound caller IDs are being displayed on my Softphones as CallingNumber@myFreePBXServerIPaddrss for example 0298765432@192.168.0.105:1234, rather than just the incoming calling number from the PSTN.

Any help would be appreciated.

Thanks

note: after any changes in certificate, you should enable/disable the UCP access in user manager. it will add new cert in sip configs.

but if you have a number of users, it would be hard.

Hi,
We have latest version of FreePBX and Fax pro module. But when we scan a paper and send it through UCP, the quality is low. We check the outgoing box and the quality of uploaded file is also low. It happens just for scanned papers. If we send original pdf or make a pdf from a word file, quality is perfect.
We also check the different qualities of scanning but result is same.
ECM is enabled too.

Do you have any idea?

Yeah, I’d really appreciate that. What screens would be best to send your way?

You just need to resave the user. You don’t have to change any settings.

Or assign all users to a group and save the group

This works on thousands of systems in the way you say it doesn’t just fine and has been working this way since version 12 which was released almost 4 years ago

Hi Avayax,

I configure following your guide, it worked, but one or two days, it will drop the calls. the error in FreePBX is "Got SIP response 603 “Declined” back from gateway.

I though because of DIGIUM device problem. I have changed another. but still same problem? did you have experience on this?

below is what i configured for Digium gateway

image.png968x572 40 KB

image.png967x363 26.2 KB

image.png654x689 41.2 KB

image.png589x724 36.4 KB

Hard to tell.
Maybe the gateway fails to send the call out the PRI.
I would do a PRI/Sip debug on the gateway and look at the information there.

Also I would look at the calls that fail if there is something specific about them. Is it calls to specific number, e.g.

(post withdrawn by author, will be automatically deleted in 24 hours unless flagged)

I am running the asterisk rvv and I can see that my global variable 'SIPDOMAIN' is set to my internal IP and not my External NAT address. I have set General SIP Settings external ip, I have set Chan PJSIP settings External IP, but none of these seem to be making a change.

hi avayax,

I have upload the debug file on https://nofile.io/f/bMEPMRu5huS/gateway_debug_asterisk_2018-05-13_08-26-34.log. Please help

There were no distro updates for 10.13.66 since November 2017 (latest kernel and package versions are from CentOS/RHEL 6.8 while 6.9 has a number of security fixes https://access.redhat.com/errata/RHSA-2017:0817) and for SNG7 since December 2017. I start thinking that if you are using FreePBX in a corporate environment it is better to use it on CentOS then a distro version for security reasons, as OS updates are coming too slowly.

Thank you for your answer i have tried all of your suggestions but it doesn’t seem to work out it shows the same message : connected (press CTRL+C to quit).
BUT
I CAN’T see this message:

== WebSocket connection from ‘127.0.0.1:37872’ for protocol ‘’ accepted using version ‘13’
Creating Stasis app ‘hello-world’