We install FreePBX-Distro-Net-1.8.1.4.iso on new server.
During the boot we are getting the following error message:
{
Setting up Logical Volume Management: /dev/hda: open failed: No medium found
}
After installation extensions registers remotely and locally successfully, however no audio is received on both end of calls at all.
Is this problem relate to “Logical Volume management”. Does the Volume word relates to sound or disk drives?
Please suggest a solution to over come the no audio problem.
The PBX is not behind any firewall, no NAT is involved.
Thanks
You provided no information at all that can be used to help.
The volume error has nothing to do with it.
If you call between two local extensions do you have audio?
You say you have no nat but you have remote extensions, does this mean the machine is directly connected to the Internet?
How about some logs?
I was trying to avoid posting my actual IP configuration on the forum publicly. Is there an email address where I can forward you the configurations?
Also the NAT problem I had been resolved. My current problem is that I installed FreePBX yesterday from scratch on a new system. This system is in a colocation site, with a static WAN IP example 75.x.x.1 and Eth0 is connected to the WAN port directly.
From remote locations phone are registering over internet to the PBX to it WAN IP directly e.g 75.x.x.1. When testing 2 extensions the voice does not get through at all on both ends.
For testing purpose I installed Elastix on the same hardware and it works fine. Extensions registered and audio works fine.
However, we prefer installing Distro over Elastix therefore would like to continue fixing the audio problem.
Looking forward for your suggestions!
Thank you
It’s still a NAT problem, just one on the remote end.
You don’t ever have to post IP’s just sanitize your output.
I am sure if you run an rtp debug from the Asterisk CLI on one of the calls with no audio you will see Asterisk is trying to send audio to the private IP of the remote NAT’d phone.
Ok, The contents of “sip_general_additional.conf” os as follows:
{
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: http://freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;
vmexten=*97
faxdetect=yes
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-2.9.0(1.8.4.2)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
}
The command “sip show seetting” returns with follwoing error:
-gasb: sip: command not found
sip show settings is an Asterisk CLI command
Here is the out put:
{
Global Settings:
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promsic. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-2.9.0(1.8.4.2)
SDP Session Name: Asterisk PBX 1.8.4.2
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Network QoS Settings:
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Jitterbuffer forced: No
Jitterbuffer max size: -1
Jitterbuffer resync: -1
Jitterbuffer impl:
Jitterbuffer log: No
Network Settings:
SIP address remapping: Disabled, no localnet list
Externhost:
externaddr: (null)
Externrefresh: 10
Global Signalling Settings:
Codecs: 0xe (gsm|ulaw|alaw)
Codec Order: ulaw:20,alaw:20,gsm:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Force rport: No
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
}
I don’t fully understand your network environment.
Did you see my RTP debug suggestion.
with your hint I looked around NAT setting and releases that in DISTRO NAT is disable by default on each extension and we have to set NAT=YES.
In Elastic NAT is YES by default that why Elastic was following the same routine.
Any ways my Audio issue is resolved with NAT=Yes.
Thank you for your tips.
I hate to say I told you so, but I told you so.
If you had listened to me you would have saved much time and aggravation.