No Sound after updates last night


(Kleinsgmbh) #1

after updating this modules

There are 12 modules available for online upgrades backup
13.0.26.6 (current: 13.0.26.5) callback 13.0.5.1 (current: 13.0.5) callerid
13.0.8.2 (current: 13.0.8.1) conferencespro 13.0.27.7 (current: 13.0.27.5) core
13.0.120.2 (current: 13.0.119.10) extensionroutes 13.0.10.4 (current:
13.0.10.3) framework 13.0.192.9 (current: 13.0.192.8) ivr 13.0.27.3 (current: 13.0.27.1)
recordings 13.0.30.11 (current: 13.0.30.9) sysadmin 13.0.74.4 (current:
13.0.74.3) tts 13.0.10 (current: 13.0.9) vqplus 13.0.26.8 (current: 13.0.26.6)

we have no sound in our phones, the phone rings, but we can NOT here the annoucing for waittime, we can NOT here the calling person.

The Caller hears the IVR, can choose something, heres music whils our phones are rining, but as soon we accept the call, there is nothing more to hear.

Please, is there a way to hotfix this?

What can i do, instead of waiting for the devs to fix it.

best regards


Debian+ freepbx 13: server broken after upgrades
All calls suddenly placed on hold, no way to recover
No outbound audio (internal and external) since last update to Edge Track
All Calls Held
(Artem Karpenko) #2

The same here. No sound between clients and in conferences. Conference admin PIN not working. IVR and music on hold - working great.

Upd:
SUPPORT ISSUE created https://issues.freepbx.org/browse/FREEPBX-15148


(Pavel Zichovsky) #3

Hello, we had same isssue after upgrade. Quick remedy is to rollback “core” module to previous version. Now sound works OK.


(Kleinsgmbh) #4

Ah, thank you for that info.

Can you tell me “how to rollback”?

I searched but diidnt find the answer, yet.


(Artem Karpenko) #5

Thank You. Rolled module ‘core’ back to 13.0.119.10 - sound came back.

‘Module admin’ -> [Check online] -> Applications-Core > Previous


(Alexandre Maurer) #6

You saved me !!!

No sound after updating to core 13.0.120.2
What a nightmare !


(Brian Bowles) #7

I experienced the same issue. Thanks for the rollback instructions.


(Kleinsgmbh) #8

Tanks, that helped me :slight_smile:

With core 13.0.119.10 it works for us, like it used to.


(Lorne Gaetz) #9

For people having audio issues, you can roll back core using the following command:

fwconsole ma downloadinstall core --tag 13.0.119.10

But we need to know what is going on here, those who are affected:

  • What Asterisk version are you running?
  • Are you seeing this on PJSIP or chan_sip extensions or both?

(Artem Karpenko) #10

Stock FreePBX 13.0.192.9
Asterisk 13.16.0
PJSIP extensions

Logs didn’t show any errors - just no sound.

Anything else?


(Kleinsgmbh) #11

Current Asterisk Version: 13.9.1
PBX Firmware: 10.13.66-11
PJSip Endpoints


(Pavel Zichovsky) #12

after core rollback:
FreePBX distro 10.13.66-20
Asterisk 13.15.0
only PJSIP endpoints and trunks.

Problem was in calls between extensions and external trunks. our users hear other side (over trunk), but other side did not hear our users. Whichever side initialized call (trunk->extension or extension->trunk), always extension heard trunk, but trunk not heard extension.


(Stacey Stemach) #13

Our FreePBX is on the following versions:
Firmware 10.13.66-20
Asterisk 13.16.0.

We only have PJSIP extensions. All extensions affected.
(Rolled back Core, resolved. Upgraded Core to 13.0.120.3, works fine.)


(Andrew Nagy) #14

Hi guys.

I want to firstly extremely apologize on behalf of Sangoma Technologies, Inc for this issue then point out what happened and finally let you know how we are going to resolve it.

Firstly this issue is related to https://issues.freepbx.org/browse/FREEPBX-15009 which is in turn related to https://issues.freepbx.org/browse/FREEPBX-14735.

Secondly what you need to know is how our QA process currently works. That is everything is merged into a single release. Our QA team then goes through and “passes” tickets by testing modules, sometimes this means there are several “fixes” or “features” in a single release. This has the potential to cause issues (as you’ve now seen).

On that a bug slipped in from 14735 which was reported in 15009 and then fixed in that ticket. However the tickets below 15009 ended up getting QA passed (because their testing was not related to this issue, however because everything is merged together the buggy code got in there) and thus merged into the code base which 15009 stayed in edge which the audio fix. We have now published 15009 as stable. This bug specifically dealt with audio addressing on PJSIP only, which broke because Asterisk does not support a media_address setting of 0.0.0.0. The fix was issued in this commit: https://github.com/FreePBX/core/commit/b5eb0cf0a2bfec10cd6775d29be37edfc5353767 which was in Core 13.0.120.3 in edge only.

So how are we going to fix this? Well we have started a new QA process as of two weeks ago (though not fully implemented) which tests each bug fix or feature request as a separate entity. This means that the QA team will be checking 1 single issue instead of a mass of issues at one time. The code they are checking won’t merge until they QA pass it. So the buggy code wouldn’t have even made it into the edge release until they pass it through.

Thank you all for you patience and understanding. I understand how frustrating this has been and perhaps our your faith in upgrading has been tainted a bit, but understand that we are trying to make this a better process.

Summary: Update to Core 13.0.120.3 which will resolve your issues.

-Andrew
Sangoma Technologies, Inc


(Stacey Stemach) #15

After updating some modules this morning, we are experiencing a serious issue with our FreePBX system.
Calls (whether incoming or outgoing, external or internal) are immediately placed on hold, and there doesn’t appear to be any way to recover them.

We have 8 Polycom VVX 500 phones, and when the call is ringing, everything looks good, but as soon as the call is answered, the display changes from displaying the called number to “Held:XXX-XXX-XXXX”, and the line shows the call on hold. X-Lite does the same.

The hold appears to be coming from something besides the device. If I establish a call between my Polycom to my cellphone, the Polycom displays “Held:CELLPHONE”, and there is silence on my cellphone. If I place the call on hold from my Polycom, I hear our FreePBX’s MOH through my cellphone.

Using XLite is the same, although XLite reports “On hold by other party.” Again, if I place that call on hold from X-Lite, I immediately hear the MOH through my cellphone.

We can call out, but get silence once the connection is established. We can receive calls, but get silence on connection. We can call the Voicemail system, but immediately get silence. The phones appear as if the calls are on hold from the other side. If we call the voicemail, the call eventually disconnects, as there apparently is no activity, even if we dial our password.

Any help would be appreciated.


(Stacey Stemach) #16

A bit more information:
When I call the FreePBX system, our IVR will pick up. Selections will route the call to the appropriate extension, and if not picked up, goes to voicemail.

I can retrieve voicemails remotely by calling in.

Calls appear to be handled by the system, however, they appear to be set on hold internally. The IP phones pick up the call, but there is only silence, and the phones appear to report being on hold, however not by the device itself.

On the Dashboard, the calls do not register in the activity graph.


(Kleinsgmbh) #17

same issue here after updating this morning

ANY HOTFIX?


(Pavel Zichovsky) #18

Try rollback “core” module to previous version. Helped us with some problems after upgrade.


(Stacey Stemach) #19

Issue Resolved. Rolled back the Core Module to previous version.

Sangoma issued an updated Core Module. Upgraded to that and the issue is resolved.


(ladiko) #20

Hi, I’m running FreePBX 13 for 1.5 years and regularly update to the Edge Track on a Ubuntu 14.04 machine. Since last nights updates there is no outbound sound from all extensions. I hear people who I call or who call me, but they don’t hear me. If i make an internal call from [email protected] to [email protected] via asterisk (192.168.1.7), both sides are silent. Same with all other phones.

Anybody has an idea what’s the reason for that?

This is what gots updated last night at Tue, 6 Jun 2017 01:02:03 UTC:

Upgradable: 
+------------+---------------+----------------+
| Module     | Local Version | Online Version |
+------------+---------------+----------------+
| callback   | 13.0.5        | 13.0.5.1       |
| core       | 13.0.119.12   | 13.0.120.2     |
| framework  | 13.0.192.8    | 13.0.192.10    |
| ivr        | 13.0.27.1     | 13.0.27.3      |
| parking    | 13.0.19.5     | 13.0.19.6      |
| recordings | 13.0.30.10    | 13.0.30.11     |
| tts        | 13.0.9        | 13.0.10        |
| userman    | 13.0.76.9     | 13.0.76.10     |
+------------+---------------+----------------+
Generating CSS...Done
Module userman successfully installed
Updating Hooks...Done
Unable to access the running directory (Permission denied).  Changing to '/' for compatibility.
.already exists
checking for pricid field ..already exists
Migrating pickup groups to named pickup groups
Migrating call groups to named call groups
Checking if trunk table migration required..not needed
Checking if privacy manager options exists..already exists
Checking for noanswer_cid field..already exists
Checking for busy_cid field..already exists
Checking for chanunavail_cid field..already exists
Checking for rvolume field..already exists
Checking for noanswer_dest field..already exists
Checking for busy_dest field..already exists
Checking for chanunavail_dest field..already exists
Checking for General Setting migrations..not needed
Deleting unused globals..done
Converting IAX notransfer to transfer if needed..updated 0000 records
deleting obsoleted record_in and record_out entries..ok
checking for dest field in outbound_routes..already exists
checking for continue field in trunks..already exists
upgrading any zap trunks to dahdi if found..ok
Checking for possibly invalid emergency caller id fields..none found
Generating CSS...Done
Module core successfully installed
Updating Hooks...Done

And this is the log of an internal call from with sip set debug on and pjsip set logger on:

https://markdownshare.com/view/9cb1b780-b054-4b42-a7b9-ab96c5338cdf