We have several servers suddenly reporting this error across several carriers after module updates today. Anyone else experiencing this?
" No compatible codecs, not accepting this offer!"
Under Asteisk SIP settings where there is the list of codecs all of the codecs that used to be listed are now gone, even though they appear to be loaded in asterisk.
PIcture of asterisk sip settings https://imgur.com/a/Pzx7sUF
uh-exp-pbx-01*CLI> core show translation
Translation times between formats (in microseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
ulaw alaw gsm g726 g726aal2 adpcm slin slin slin slin slin slin slin slin slin lpc10 g729 ilbc g722 testlaw
ulaw - 9150 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 15000 17250 15000
alaw 9150 - 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 15000 17250 15000
gsm 15000 15000 - 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 15000 17250 15000
g726 15000 15000 15000 - 15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 15000 17250 15000
g726aal2 15000 15000 15000 15000 - 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 15000 17250 15000
adpcm 15000 15000 15000 15000 15000 - 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 15000 17250 15000
slin 6000 6000 6000 6000 6000 6000 - 8000 8000 8000 8000 8000 8000 8000 8000 6000 6000 6000 8250 6000
slin 14500 14500 14500 14500 14500 14500 8500 - 8000 8000 8000 8000 8000 8000 8000 14500 14500 14500 14000 14500
slin 14500 14500 14500 14500 14500 14500 8500 8500 - 8000 8000 8000 8000 8000 8000 14500 14500 14500 6000 14500
slin 14500 14500 14500 14500 14500 14500 8500 8500 8500 - 8000 8000 8000 8000 8000 14500 14500 14500 14500 14500
slin 14500 14500 14500 14500 14500 14500 8500 8500 8500 8500 - 8000 8000 8000 8000 14500 14500 14500 14500 14500
slin 14500 14500 14500 14500 14500 14500 8500 8500 8500 8500 8500 - 8000 8000 8000 14500 14500 14500 14500 14500
slin 14500 14500 14500 14500 14500 14500 8500 8500 8500 8500 8500 8500 - 8000 8000 14500 14500 14500 14500 14500
slin 14500 14500 14500 14500 14500 14500 8500 8500 8500 8500 8500 8500 8500 - 8000 14500 14500 14500 14500 14500
slin 14500 14500 14500 14500 14500 14500 8500 8500 8500 8500 8500 8500 8500 8500 - 14500 14500 14500 14500 14500
lpc10 15000 15000 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 - 15000 15000 17250 15000
g729 15000 15000 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 - 15000 17250 15000
ilbc 15000 15000 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 - 17250 15000
g722 15600 15600 15600 15600 15600 15600 9600 17500 9000 17000 17000 17000 17000 17000 17000 15600 15600 15600 - 15600
testlaw 15000 15000 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 15000 17250 -
We are also seeing this issue, but so far only on FreePBX 13 systems. Trying to narrow down which module caused the issue.
It seems to be affecting only FreePBX 13 with us as well.
With most of our carriers we set disallow=all and allow=ulaw on our trunks. Removing disallow=all and changing allow=all seems to fix the issue.
This is affecting every PBX 13 server we have with every carrier we have.
I am not sure if this is relevant but we just started to get this error on these servers today.
[2018-05-30 08:32:31] ERROR[59151] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:32:31] ERROR[59132] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:32:31] ERROR[59132] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:32:31] ERROR[6105] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:32:31] ERROR[6105] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:32:31] ERROR[59132] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:33:42] ERROR[61000] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:33:42] ERROR[60984] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:33:42] ERROR[60984] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:33:42] ERROR[6105] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:33:42] ERROR[6105] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:33:42] ERROR[60984] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:44:55] ERROR[5389] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:44:55] ERROR[5382] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:44:55] ERROR[5382] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:44:55] ERROR[5382] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:44:55] ERROR[6105] phone_message.c: Unable to build dialplan routing - invalid license
[2018-05-30 08:44:55] ERROR[6105] phone_message.c: Unable to build dialplan routing - invalid license
Just thought I would through this out there too. We are getting this error on every FreePBX 14 server we have after module updates.
exit: 1
Unable to continue. Unable to locate the FreePBX BMO Class 'Core’A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install core 2) fwconsole ma enable core in /var/www/html/admin/libraries/BMO/Self_Helper.class.php on line 213
#0 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(106): FreePBX\Self_Helper->loadObject(‘Core’)
#1 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(37): FreePBX\Self_Helper->autoLoad(‘Core’)
#2 /var/www/html/admin/modules/certman/Certman.class.php(39): FreePBX\Self_Helper->__get(‘Core’)
#3 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(124): FreePBX\modules\Certman->__construct(Object(FreePBX))
#4 /var/www/html/admin/libraries/BMO/Self_Helper.class.php(37): FreePBX\Self_Helper->autoLoad(‘Certman’)
#5 /var/www/html/admin/libraries/BMO/Hooks.class.php(294): FreePBX\Self_Helper->__get(‘Certman’)
#6 /var/www/html/admin/libraries/BMO/Hooks.class.php(39): FreePBX\Hooks->preloadBMOModules()
#7 /var/lib/asterisk/bin/retrieve_conf(81): FreePBX\Hooks->updateBMOHooks()
#8 {main}
In Self_Helper.class.php line 213:
Unable to locate the FreePBX BMO Class 'Core’A required module might be dis
abled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma
install core 2) fwconsole ma enable core
This is the command I use to update. fwconsole ma --quiet updateall
Shouldn’t this catch all of the modules?
Yes but when you --quiet it you dont see the errors. You are basically ignoring errors.
It appears that modules that were not there previously were attempted to be installed Sangomcrm and Zulu.
Right now Core is disabled. You need to fix that first.
Core is enabled.
]# fwconsole ma enable core
The following error(s) occured:
- Module core is already enabled
This is an issue on the FreePBX 14 servers. What about the Codec issue on the FreePBX13 servers?
I dont see any error on FreePBX 13 at this time with codecs. The way you see codecs is with “core show codecs audio”
Are you sure your modules are all updated. It seems like you havent installed something.
Here is the output from one of the servers with the codec issue.
asterisk]# fwconsole ma updateall
No repos specified, using: [standard,commercial,extended] from last GUI settings
Up to date.
Updating Hooks…Done
[root@uh-exp-pbx-01 asterisk]#
uh-exp-pbx-01*CLI> core show codecs audio
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
ID TYPE NAME DESCRIPTION
5 audio g726 (G.726 RFC3551)
3 audio alaw (G.711 a-law)
1 audio g723 (G.723.1)
19 audio speex (SpeeX)
20 audio speex (SpeeX 16khz)
21 audio speex (SpeeX 32khz)
23 audio g722 (G722)
7 audio adpcm (Dialogic ADPCM)
24 audio siren7 (ITU G.722.1 (Siren7, licensed from Polycom))
27 audio g719 (ITU G.719)
18 audio g729 (G.729A)
8 audio slin (16 bit Signed Linear PCM)
9 audio slin (16 bit Signed Linear PCM (12kHz))
10 audio slin (16 bit Signed Linear PCM (16kHz))
11 audio slin (16 bit Signed Linear PCM (24kHz))
12 audio slin (16 bit Signed Linear PCM (32kHz))
13 audio slin (16 bit Signed Linear PCM (44kHz))
14 audio slin (16 bit Signed Linear PCM (48kHz))
15 audio slin (16 bit Signed Linear PCM (96kHz))
16 audio slin (16 bit Signed Linear PCM (192kHz))
2 audio ulaw (G.711 u-law)
17 audio lpc10 (LPC10)
26 audio testlaw (G.711 test-law)
39 audio none (<Null> codec)
25 audio siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
6 audio g726aal2 (G.726 AAL2)
4 audio gsm (GSM)
22 audio ilbc (iLBC)
28 audio opus (Opus Codec)Did you see my screen shot of the asterisk sip settings where the codec list is not correct?
Yes. Everything works fine for me. If you go to Asterisk SIP Settings you dont see any codecs?
