Sip setting new upgrade to 12.0.10

Upgrade module from sip setting 12.0.9 to 12.0.10 is not working.
Error message : Error(s) downloading sipsettings: Unable to work with GPG file, message was: I can only do .gpg files at the moment.

I’m getting the same. I was wondering if I was the only one.

this is the size of one that works

-rwxrwxr-x 1 asterisk asterisk 96916 Mar 10 00:57 sipsettings-12.0.10.tgz.gpg

this is the size of one that doesn’t (current)

-rwxrwxr-x 1 asterisk asterisk 96904 Mar 10 19:20 sipsettings-12.0.10.tgz.gpg

Something’s changed from the download site.

I was able to manually install the copy I got from last nite and it works. Definitely something corrupt on the DL site.

Whoops, 100% my fault. I regenerated the tar, but didn’t update the checksum. You may all hurl rotten fruit at me. Sorry!

Thanks a lot for your help Andrew!

That command gave me some erros about “undefined function posix_getpwuid()” but I was able to fix it by installing php-posix and then running “amportal chown” as suggested, that fixed the interface problem…

Now, with the upgrade to 12: I was able to install the Upgrader Module, upgrade Framework and Core but when upgrading all the rest I get this error on the sipsettings module (all other modules download and install fine):

Error(s) downloading sipsettings: File Integrity failed for /var/www/html/admin/modules/_cache/sipsettings-12.0.10.tgz - aborting (md5sum did not match)

Then, if I try that module again I then start getting this error:

Downloading and Installing sipsettings
Downloading sipsettings 96904 of 96904 (100%)
Found module locally, verifying…Redownloading
Error(s) downloading sipsettings: Unable to work with GPG file, message was: I can only do .gpg files at the moment

I thought maybe a download error may have caused the problem so I reverted to a snapshot just before the upgrade to 12 (this is a virtual server) and started again, exactly the same happened with sipsettings module…

could there be a problem with that file on the mirror I’m hitting? or a problem with my install?

Thanks!

EDIT: if I delete the sipsettings-12.0.10.tgz.gpg file from the _cache folder then I get the “md5sum did not match” error again…

When I upgrade to this new module, I immediately lose connectivity - calls go through but the minute the called party answers, the call is disconnected. To test this I set up an extension using a softphone (Telephone on OS X). Before upgrading to the module, I can place calls easily to another number on an interconnected PBX, and the calls go straight through with no issues.

Then I upgrade to the new module and try the exact same call again. The phone on the interconnected PBX still rings, but the moment it answers, I get firewall popups asking me to approve certain ports, and once I do that, the call immediately disconnects. Other users on the system also report that their calls will not go through - the called phone rings, but when the called party answers the call immediately disconnects.

I have before and after copies of the /etc/asterisk directories and by running a diff comparison tool I find that only two files are modified by the upgrade. One is extensions_additional.conf, which in the [globals] section of the file gets an added line:

SIPLANG =

This is added right at the end of the [globals] section, on line 422 of the file.

The major changes are in the sip_general_additional.conf file. Prior to the module upgrade, it looks like this:

;--------------------------------------------------------------------------------;
;          Do NOT edit this file as it is auto-generated by FreePBX.             ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate.                                                         ;
;--------------------------------------------------------------------------------;
accept_outofcall_message=yes
auth_message_requests=no
outofcall_message_context=dpma_message_context
faxdetect=no
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-12.0.43(11.16.0)
disallow=all
allow=ulaw
allow=g722
allow=alaw
allow=gsm
allow=g726
bindport=5060
localnet=xx.xx.xx.xx/24

But after upgrading the module, several new lines are added:

;--------------------------------------------------------------------------------;
;          Do NOT edit this file as it is auto-generated by FreePBX.             ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate.                                                         ;
;--------------------------------------------------------------------------------;
accept_outofcall_message=yes
auth_message_requests=no
outofcall_message_context=dpma_message_context
faxdetect=no
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-12.0.43(11.16.0)
disallow=all
allow=ulaw
allow=g722
allow=alaw
allow=gsm
allow=g726
callevents=no
rtpstart=10000
rtpend=20000
bindport=5060
jbenable=no
allowguest=yes
srvlookup=no
defaultexpiry=120
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=20
g726nonstandard=no
videosupport=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
checkmwi=10
notifyringing=yes
notifyhold=yes
nat=yes
ALLOW_SIP_ANON=no
externip=xx.xx.xx.xx
localnet=xx.xx.xx.xx/24

Specifically, the following lines are ADDED:

callevents=no
rtpstart=10000
rtpend=20000
jbenable=no
allowguest=yes
srvlookup=no
defaultexpiry=120
minexpiry=60
maxexpiry=3600
registerattempts=0
registertimeout=20
g726nonstandard=no
videosupport=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
checkmwi=10
notifyringing=yes
notifyhold=yes
nat=yes
ALLOW_SIP_ANON=no
externip=xx.xx.xx.xx

It appears no existing lines were changed, so it must be something in that batch of added lines that is causing the problem. Those are the only two files that are changed in the entire /etc/asterisk directory. I just want things to work as they did before upgrading the module. Or, alternately, to be able to not upgrade the module but not get an email every day nagging me about it, while still getting notified of other module upgrades.

As soon as I restore a snapshot made just prior to upgrading the module, everything works great again. Any idea why calls would stop working when I upgrade this module?

So there’s actually no issue with the upgrade of the module. What did happen was that we fixed it so that it will write out all default values as they are shown in SIP Settings itself. Which is why you are seeing lines added. What you should do is go into sip settings and edit the values so until it works so that you can get updates in the future.

Also why restore a complete backup when you can do module rollback?

Sorry, there is no way to do that.

Sorry, there is no way to do that.
[/quote]

Is there a way to change the notification frequency? Seems like in the past I used to only get these notifications once a week instead of every day.

Or to approach this from another direction, what is the relationship of sip_general_additional.conf to sip_general_custom.conf? If I copied the settings that I know work into sip_general_custom.conf, would it override the settings in sip_general_additional.conf, or do the ones in sip_general_additional.conf take priority?

It only shows up after you’ve “check online”

No

You won’t be able to do that. Looks like maybe your nat settings need to be changed. But all of these settings exist inside of Sip Settings. You could also remove each new line added and reload just asterisk until you found the line that is causing the issue.