Hi all! After install i lost web interface.

HI

I'm checking the latest 32bit USB image available
(Stable-5.211.65-1) just downloaded

I'm flashing a 8GB usb pen the usual way:
dd if=distro.img of=/dev/sdx

There is no way to make it boot (it says boot error on more PC)

Just flashed the 4.211.64-7 the same way , it works like a charm.

Is there any known issue with the usb image ??

At the moment I'm dowloading 4.211.64-9 and 5.211.65-1(64bit)
to see how those behave......

Thanks

I have an extensive IT background, but I'm new to FreePBX and still learning CentOS. I have FreePBX 2.11.0.11 running on CentOS 6.4 (all installed via PIAF2 2.0.6.4.5 64BIT). I have reloaded the whole machine with the same results. Working with my ISP and SIP trunk provider who "Doesn't support Asterisk due to the many different flavours available", I know the account work by setting it up on an X-Lite softphone and a Yealink IP handset, and it works on both. As for my FreePBX install, I can have extensions call each other over my LAN with perfect results. Any extension can call out to a PSTN phone number, and the call is perfect. The trouble is, I can not recieve any inbound calls. The phone dialing in recieves a "Fast Busy" signal. Checking with the SIP Trunk provider, they are not seeing any registration attempts from my IP (Static Public IP with the proper inbound port forwarding configured). They provided me with "Asterisk setup advice" as per other clients that have set it up, however they are not directly supporting me on this.
Here is their "advice"...
..................................................
The Asterisk PBX can and is being used on the Galaxy network but due to the myriad of permutations with varying versions, OS support issues and customized configurations, Asterisk is not officially supported.

However, the following example sip.conf is provided to help get started (based on the example VoIP number: 200112345678 with password: abc123):

------

[general]
srvlookup=yes

register => 200112345678:abc123@200112345678/200112345678
registertimeout=60
registerattempts=0

[authentication]
[200112345678]
type=peer
secret= abc123
username=200112345678
host=ep.asterisk.rgns.net
fromuser=200112345678
outboundproxy=ep.asterisk.rgns.net:15061
insecure=invite
context=default
disallow=all ; Note: In user sections the order of codecs listed with allow= does NOT matter!
allow=ulaw
allow=g729
qualify=yes
qualifyfreq=40

------

The @200112345678 portion of the register line references the context [200112345678]. The settings under [200112345678] will be used for registration and in turn the same settings under [200112345678] can be used to for outbound calls, sourced from extensions.conf. The appending /200112345678 of the register line is the extension, this will allow the Asterisk box to register with a Contact of 200112345678 and in turn you can create an extension of 200112345678 in extension.conf to receive calls, ie:

[default]
exten => 200112345678,1,Goto(some_extension,1,1)

[some_extension]
exten => 1,1,Wait(1)
exten => 1,2,NoOp(${CALLERID(all)})
exten => 1,3,background(/var/lib/asterisk/sounds/greeting)
exten => 1,4,Dial(SIP/101,20)

Note: If asterisk has difficulty looking up DNS SRV records, try changing the outboundproxy to: outboundproxy=ep.asterisk.rgns.net:15061
.................................................
So I have been able to rule it down to my FreePBX box being the issue. I have tried all of this from my network, as well as I moved the PBX box and phones to my local ISP's office, where they let me troubleshoot at a desk there. I know that it isn't a network config issue... it is something in the FreePBX setup.
My SIP Trunk provider has not seen any traffic from my IP to their's trying to register. Today, I SSH'ed into my box and ran a "sip reload" which came up with...

[2013-12-18 12:31:11] WARNING[1785]: sip/config_parser.c:132 sip_parse_register_line: Format for registration is [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] at line 25

[2013-12-18 12:31:11] WARNING[1785]: sip/config_parser.c:812 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead

[2013-12-18 12:31:11] WARNING[1785]: sip/config_parser.c:132 sip_parse_register_line: Format for registration is [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension]
[~expiry] at line 8

[2013-12-18 12:31:11] SECURITY[1749]: res_security_log.c:134 security_event_cb:SecurityEvent="RequestBadFormat",EventTV="1387387871-530801",Severity="Error",Serrvice="AMI",EventVersion="1",SessionID="0x28da5f8",LocalAddress="IPV4/TCP/0.0.0.0/5038",RemoteAddress="IPV4/TCP/127.0.0.1/42201",RequestType="Action: ZapShowChannels",SessionTV="1386679247-879539",AccountID="admin

I'm sure there is enough info here to pick out some mistake that I have made. Just to recap... Extension to extension calls are perfect. Extension out on the trunk works as it is listed as a "PEER", and the inbound calls simply get a fast busy on their phone, which corresponds to my SIP Trunk provider not having a client registered to that account. Using a SIP phone direct to my SIP trunk provider works and registers just fine. It really seems to boil down to FreePBX simply not even trying to register. Any help would be great. This is driving me to drink!

Hey all!

Hopefully someone can point me in the right direction...

Currently I have a system setup using AsteriskNOW and 13 Digium D50's. When a call comes in it goes to a goes to a time condition, if yes it goes to the receptionist, if no it goes to a ring group that rings all phones in the office as there are often people there after the receptionist leaves.

So currently if she leaves her desk she has to dial the time condition override, however there is no visual status of this on the phone.

Ideally what I would like would be to figure out a way to ring her phone if its set to Available but if its set to DND or Away, it then goes on to the ring group.
I just cant figure out a good way to do that, although I'm sure there is one.

I have tried to search but perhaps I am using the wrong terms because I cant seem to find anything

Thanks

-Jeff

Hi guys,

First of all, i'd like to inform you that i'm new user of Asterisk and Freepbx. Sorry for my questions if they seems not making sense sometimes .

I just succefully installed AsteriskNow+Freepbx on my home Laptop following this user guide (https://wiki.asterisk.org/wiki/display/ ... steriskNOW). Now i'm trying to configure the freepbx by pointing my web browser to http:/10.0.2.15. But it does'nt work on my IE or Chrome browsers.

Could you help me please ? Thanks.

My environnement configuration:
- AsteriskNow installed on a VirtualBox machine
- Dell 32 Bytes laptop
- Windows Vista OS

Hi,

I have a VMBlast Group setup as an option in an IVR. When callers select that option, the audio label plays but then this "If this is correct, press 1" comes up. Can I get rid of it and just have it go right to the beep and recording?

Thanks,
Sam

Hello!
I installed FreePBX 2.11.0.10 on my Raspberry Pi. I want change language in WebGUI to Spanish (for example) but it isn't possible. When i clicked to other language the page was refreshed but language wasn't changed.

I tried this methods:
http://www.freepbx.org/forum/freepbx/installation/change-the-language
...But it wasn't helped me.

If you don't know how to fix this problem I have next question...Where is saved the default language (en_US) file? I can't find it in "/var/www/html/admin/i18n".

Please, help me with this problem. Thank you.

P.S.: I'm sorry for my English, I'm from Czech.

Starting with Cent OS 6.5 the Minimal install CD seems to be also fully EFI compliant. Therefore I try again a native EFI install on my Mac-mini. I have downloaded the latest Cent OS 6.5 based FreePBX-5.211.65-3-x86_64 build.

Well, to make it short, it failed again! It failed in exactly the same way like all previous (EFI modded) FreePBX-4.x install attempts.

The symptom is always identical, - after booting from install media, there is visible for a few seconds a "scrambled, distorted screen". (Probably the wrong configured FreePBX grub-splash.xpm.gz image?) Then the whole installation is "falling back" immediately to stock Cent OS 6.5 install mode which is displayed then correctly.

After searching around the web I found a very interesting bug report on redhat bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=820680

That sounds all very familiar, could it be possible that all these FreePBX Kickstart files / Kickstart sourcefiles are build somehow "EFI incompatible"?

Try to check it m myself, - was unable to find any "--fstype" or "bootloader --location=partition --driveorder=sda" parameters. My knowledge regrading all this is Linux stuff is too limited...

Hello Community

I have successfully installed and configured FreePBX (2.11.0.11). Due to network reconfiguration, we had to change the IP and we have installed a new internet line dedicated for FreePBX. Since then, we cannot receive inbound calls anymore.

- Originally, my server was:192.168.2.30, Subnet:255.255.255.0, Def gateway: 192.168.2.1. I had a fix IP address (which was my WAN IP).
- I reconfigured the server to be on 192.168.3.30, sub: 255.255.255.0, DefGW: 192.168.3.1. New internet line with New fix IP address.
- I reconfigured the routes on my firewall to point to the new IP..
- Cannot receive Inbound calls.

What did I missed?

Tks

Daniel

Hi, i've downloaded and installed latest asterisknow with Freepbx.
I configure (hope correctly) my Isdn Pci card:
[dahdi_hardware command]
pci:0000:01:0e.0 zaphfc+ 1397:2bd0 HFC-S ISDN BRI card

[dahdi_scan command]
[1]
active=yes
alarms=OK
description=HFC-S PCI A ISDN card 0 [TE]
name=ZTHFC1
manufacturer=Cologne Chips
devicetype=HFC-S PCI-A ISDN
location=PCI Bus 01 Slot 15
basechan=1
totchans=3
irq=0
type=digital-TE
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=AMI
framing_opts=CCS
coding=AMI
framing=CCS

[dahdi-channels.conf]
; Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] " (MASTER) AMI/CCS
group=0,11
context=from_dahdi
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel => 1-2
context = default
group = 63

[chan_dahdi.conf]
[channels]
language=it
busydetect=yes
busycount=10
usecallerid=yes
callwaiting=yes
usecallingpres=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=no
immediate=no
faxdetect=yes
rxgain=0.0
txgain=0.0

In asterisk:
[dahdi show status]
Description Alarms IRQ bpviol CRC Fra Codi Options LBO
HFC-S PCI A ISDN card 0 [TE] OK 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1)

[dahdi show channels]
Chan Extens Context Lang MOH Interpret Blocked State Descr.
pseudo default default In Service
1 from-dahdi it default In Service
2 from-dahdi it default In Service

when i try to call in asterisk -vvvvvvvv i see this message:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [0426xxxxxx@from-internal:1] ResetCDR("SIP/101-0000000a", "") in new stack
-- Executing [0426xxxxxx@from-internal:2] NoCDR("SIP/101-0000000a", "") in new stack
-- Executing [0426xxxxxx@from-internal:3] Progress("SIP/101-0000000a", "") in new stack
-- Executing [0426xxxxxx@from-internal:4] Wait("SIP/101-0000000a", "1") in new stack
> 0xb5c2c988 -- Probation passed - setting RTP source address to 192.168.1.31:16480
-- Executing [0426xxxxxx@from-internal:5] Progress("SIP/101-0000000a", "") in new stack
-- Executing [0426xxxxxx@from-internal:6] Playback("SIP/101-0000000a", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- Playing 'silence/1.slin' (language 'it')
-- Playing 'cannot-complete-as-dialed.gsm' (language 'it')
-- Playing 'check-number-dial-again.gsm' (language 'it')
-- Executing [0426xxxxxx@from-internal:7] Wait("SIP/101-0000000a", "1") in new stack
-- Executing [0426xxxxxx@from-internal:8] Congestion("SIP/101-0000000a", "20") in new stack
[2014-01-02 13:20:46] WARNING[2084][C-0000000b]: channel.c:4816 ast_prod: Prodding channel 'SIP/101-0000000a' failed
== Spawn extension (from-internal, 0426xxxxxx, exited non-zero on 'SIP/101-0000000a'
-- Executing [h@from-internal:1] Hangup("SIP/101-0000000a", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-0000000a'

Incoming call are all ok but i can make outgoing call.
What's wrong?

hello,

just wanna know if freePBX will run smoothly on the below platform. it will just be use on a maximum of 15 users environment;

= pc engines alix boards (alix3d2 = 1 LAN / 2 miniPCI / LX800 / 256 MB with 8GB SLC CF card)
= cisco 3102 for my gateway
= and planning to add an external usb flash disk for logs.

i would appreciate any suggestions/recommendations using embedded platforms on such a number of users.

thx

Greetings,

I am having trouble with recordings under the user panel. I tried to use some of the information from this thread:
http://www.freepbx.org/forum/beta-program-issues/user-panel-not-accessable-on-freepbx-211-beta

I know the user panel button is not directing to correct location, and i believe that is part of the problem. I believe asterisk is recording but I cannot pull information up. is there a way to modify the web text and update the file path.

Thanks,
Mac

I'm totally new with VOIP and SIP, so please be patient! I have a Freepbx box set up, and am using Csipsimple on my Android to use as an extension phone. The problem is that I get the error message "error when registering - not implimented" on the phone. How can I trouble shoot this?

It did register for a while, but no longer does.

Thanks for the help.

James

hi all is it possible to install freepbx on TBS 2910 Matrix..

http://www.redappledigital.co.uk/Matrix-ARM-mini-PC-TBS-2910-p/tbs2910.htm#

By raspi has faild no power going to it have checked psu and its the raspi its self... i would like quad core as with its better for me as we can run our rf radio gateway on the same device!

Total newb to Linux and FreePBX.

I downloaded and installed the latest PIAF distro. (CentOS 6.5, Asterisk 1.8.25, FreePBX 2.11.0.12), on a PC.
I was able to set up a softphone, but when I try to set up Google for trunk, I select "Connectivity" tab and Google Voice(Motif), I get the message "This module requires Asterisk chan_motif & res_xmpp to be installed and loaded".
I went to Module Admin and updated everything I could find.
I tried the asterisk-version-switch from the command line that I found online, but received "command not found".

I'm stumped (of course).
Any help would be gratefully appreciated.

I just bought a new server from Dell and an 8-port analog telephony card from Digium. Upon installing AsteriskNow with FreePBX the analog card was not recognized in FreePBX.

Following the suggestions of Digium support and some forum users, I performed a CentOS update using yum update and updated the core module in FreePBX.

The card is now recognized and shows that DAHDI version 2.8.0.1 is now being used. The only problem is, that is sees them as FXS ports instead of FXO ports.

I confirmed with Digium support that Asterisk is seeing them as FXO ports.

How do I get FreePBX to see the ports correctly?

Howdy community,

I'm looking for some assistance with a issue that I have not been able to call out when the CID is outside the nominal standard extension used by the ITSP

Software:
FreePBX 2.11.0
Asterisk 11.6.0

Description:
After discussions with our ITSP, we have moved from hundreds of SIP trunks, one per extension, to just 1 trunk with multiple CIDs outgoing. Add to this, we have a ISDN number range from another Telco, and the ITSP has agreed to allow this as the we can provide evidence of ownership.

Our problem is that if we send a From: that is from the ISDN number range, the call is rejected by the ITSP as unauthorised.

Discussions with the ITSP have indicated that we dont have the correct header information.

We must have a static header containing the Primary account number (in this case the Trunk authorised username) in the Contact: field.

eg:
From: ;tag=as7b65a2e1
To:
Contact:

So the From and To fields are dynamic, but the Contact field is static.

Makes no difference with the Sip settings NAT configuration.

Any help would be appreciated.

I had done installation and configure trunk and in/out call option but still incoming and outgoing not working.

== Starting D-Channel on span 1
== Registered channel type 'DAHDI' (DAHDI Telephony Driver w/PRI & SS7 & MFC/R2)
== Registered application 'DAHDIAcceptR2Call'
== Manager registered action DAHDITransfer
== Manager registered action DAHDIHangup
== Manager registered action DAHDIDialOffhook
== Manager registered action DAHDIDNDon
== Manager registered action DAHDIDNDoff
== Manager registered action DAHDIShowChannels
== Manager registered action DAHDIRestart
== Manager registered action PRIShowSpans
Loaded chan_dahdi.so => (DAHDI Telephony Driver w/PRI & SS7 & MFC/R2)
-- Remote UNIX connection disconnected
== Primary D-Channel on span 1 up
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [99900039877@from-internal:1] ResetCDR("SIP/39877-00000014", "") in new stack
-- Executing [99900039877@from-internal:2] NoCDR("SIP/39877-00000014", "") in new stack
-- Executing [99900039877@from-internal:3] Progress("SIP/39877-00000014", "") in new stack
-- Executing [99900039877@from-internal:4] Wait("SIP/39877-00000014", "1") in new stack
> 0x7f69f4288610 -- Probation passed - setting RTP source address to 192.168.10.76:5004
-- Executing [99900039877@from-internal:5] Progress("SIP/39877-00000014", "") in new stack
-- Executing [99900039877@from-internal:6] Playback("SIP/39877-00000014", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- Playing 'silence/1.ulaw' (language 'en')
-- Playing 'cannot-complete-as-dialed.gsm' (language 'en')
== Spawn extension (from-internal, 99900039877, 6) exited non-zero on 'SIP/39877-00000014'
-- Executing [h@from-internal:1] Hangup("SIP/39877-00000014", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/39877-00000014'

Hello,

I'm new to the PBX world, so please be patient with me. I just installed the latest copy of the distro onto an old trixbox server with a sangoma a200 FXO setup. After the install completes (for the 4th time) I can access the GUI fine and most of the features however, i'm missing the System Admin to set network settings and other modules such as the Endpoint manager etc... I installed the distro on my Mac using VM and all the modules appeared here. Any ideas on whats happening? When installing the distro i select 'Full install'. I also get a first time boot error telling me to check my internet connection, yet i'm still connected.

Thanks!
Noah

Hello Guys,

I am having problem with E1 connection.

My setup : freepbx 2.11.011 (asterisk 1.8) with Sangoma B601 card. Enabled dadhi on web panel and setup card using setup-sangoma.

Wanrouter status
Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate |
wanpipe2 | N/A | A200/A400/B600/B700/B800/B610| 17 | 4 | 1 | N/A | 0 |
wanpipe1 | N/A | A101/1D/2/2D/4/4D/8/8D/16/16D| 17 | 4 | 1 | N/A | 0 |

Wanrouter Status:

Device name | Protocol | Station | Status |
wanpipe2 | A-ANALOG | N/A | Connected |
wanpipe1 | AFT TE1 | N/A | Connected |

Wanpipe show connected but when i logon to asterisk i am getting below error.

pri set debug on span 1
Enabled debugging on span 1
PRI Span: 1 TEI=0 Got SABME from network peer.
PRI Span: 1 TEI=0 Sending UA
PRI Span: 1 TEI=0 MDL-ERROR (F): SABME in state 7(Multi-frame established)
PRI Span: 1 TEI=0 Got SABME from network peer.
PRI Span: 1 TEI=0 Sending UA
PRI Span: 1 TEI=0 MDL-ERROR (F): SABME in state 7(Multi-frame established)
PRI Span: 1 TEI=0 Got SABME from network peer.
PRI Span: 1 TEI=0 Sending UA
PRI Span: 1 TEI=0 MDL-ERROR (F): SABME in state 7(Multi-frame established)
PRI Span: 1 TEI=0 Got SABME from network peer.
PRI Span: 1 TEI=0 Sending UA
PRI Span: 1 TEI=0 MDL-ERROR (F): SABME in state 7(Multi-frame established)
PRI Span: 1 TEI=0 Got SABME from network peer.
PRI Span: 1 TEI=0 Sending UA
PRI Span: 1 TEI=0 MDL-ERROR (F): SABME in state 7(Multi-frame established)
PRI Span: 1 TEI=0 Got SABME from network peer.
PRI Span: 1 TEI=0 Sending UA
PRI Span: 1 TEI=0 MDL-ERROR (F): SABME in state 7(Multi-frame established)
PRI Span: 1 TEI=0 Got SABME from network peer.
PRI Span: 1 TEI=0 Sending UA
PRI Span: 1 TEI=0 MDL-ERROR (F): SABME in state 7(Multi-frame established)
PRI Span: 1 TEI=0 Got SABME from network peer.
PRI Span: 1 TEI=0 Sending UA
PRI Span: 1 TEI=0 MDL-ERROR (F): SABME in state 7(Multi-frame established)
PRI Span: 1 TEI=0 Got SABME from network peer.
PRI Span: 1 TEI=0 Sending UA
PRI Span: 1 TEI=0 MDL-ERROR (F): SABME in state 7(Multi-frame established)

/etc/dahdi/system.conf

system.conf
span=1,1,0,CCS,HDB3,CRC4
bchan=1-15,17-31
dchan=16
fxsks=32-35
fxoks=36
loadzone=nz
defaultzone=nz

/ete/asterisk/dahdi-channels.conf

; Span 1: WPE1/0 "wanpipe1 card 0" YELLOW RED
group=0,11
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31
context = default
group = 63

; Span 2: WRTDM/0 "wrtdm Board 1" (MASTER)
;;; line="32 WRTDM/0/0"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 32
callerid=
group=
context=default

;;; line="33 WRTDM/0/1"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 33
callerid=
group=
context=default

;;; line="34 WRTDM/0/2"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 34
callerid=
group=
context=default

;;; line="35 WRTDM/0/3"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 35
callerid=
group=
context=default

;;; line="36 WRTDM/0/4"
signalling=fxo_ks
callerid="Channel 36"<4036>
mailbox=4036
group=5
context=from-internal
channel => 36
callerid=
mailbox=
group=
context=default

/etc/wanpipe/wanpipe.1

[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE = AFT
S514CPU = A
CommPort = PRI
AUTO_PCISLOT = NO
PCISLOT = 4
PCIBUS = 3
FE_MEDIA = E1
FE_LCODE = HDB3
FE_FRAME = CRC4
FE_LINE = 2
TE_CLOCK = NORMAL
TE_REF_CLOCK = 0
TE_SIG_MODE = CCS
TE_HIGHIMPEDANCE = NO
TE_RX_SLEVEL = 430
HW_RJ45_PORT_MAP = DEFAULT
LBO = 120OH
FE_TXTRISTATE = NO
MTU = 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO
TDMV_SPAN = 1
TDMV_DCHAN = 16
TE_AIS_MAINTENANCE = NO #NO: defualt YES: Start port in AIS Blue Alarm and keep line down
#wanpipemon -i w1g1 -c Ttx_ais_off to disable AIS maintenance mode
#wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode
TDMV_HW_DTMF = NO # YES: receive dtmf events from hardware
TDMV_HW_FAX_DETECT = NO # YES: receive fax 1100hz events from hardware
HWEC_OPERATION_MODE = OCT_NORMAL # OCT_NORMAL: echo cancelation enabled with nlp (default)
# OCT_SPEECH: improves software tone detection by disabling NLP (echo possible)
# OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions.
HWEC_DTMF_REMOVAL = NO # NO: default YES: remove dtmf out of incoming media (must have hwdtmf enabled)
HWEC_NOISE_REDUCTION = NO # NO: default YES: reduces noise on the line - could break fax
HWEC_ACUSTIC_ECHO = NO # NO: default YES: enables acustic echo cancelation
HWEC_NLP_DISABLE = NO # NO: default YES: guarantees software tone detection (possible echo)
HWEC_TX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default)
HWEC_RX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default)
HWEC_TX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal
HWEC_RX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal

[w1g1]
ACTIVE_CH = ALL
TDMV_HWEC = YES
MTU = 8

Telco is telling me that every thing is fine on their end and there is an issue with my setup.
Please help i cant figured out how to fix this issue. I have tried re configuring card but no luck.

Brian