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Type:
Bug
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Status: Closed
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Priority:
Minor
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Resolution: Duplicate
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Affects Version/s: 13
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Fix Version/s: 13
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Component/s: FreePBX Framework
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Labels:None
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ToDo:
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Asterisk Version:tested with 13.6.0, 14.4.0 and 14.5.0
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Distro Version:14.04.5
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Distro:Self Install Ubuntu
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 74@192.168.1.8 to 78@192.168.1.125 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`:
> Link deleted the log as i found the bug.
- is duplicated by
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FREEPBX-15148 No Sound after updates last night
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- Closed
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FREEPBX-15178 Received calls are "held"
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- Closed
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- relates to
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FREEPBX-14735 RTP sent from wrong IP Address in HA Cluster when using PJSip
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- Closed
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