@gridsuk wrote:

Hi There,

I have been trying for a number of hours to get the asterisk mini-HTTP server to bind on TCP port 443. I have changed the port in the config but it does not bind when I look at netstat. If I change this port to 8089 it binds fine.

I did some testing and managed to get it to bind on ports 1024 and above so I am guessing this is a problem with permissions binding to the port. Can anyone please offer any advice on how I can get this working?

I have moved existing admin secure ports to none default ports so there is nothing currently listening on there.

Many Thanks

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@N1ck_J wrote:

Good afternoon,

Can anyone confirm the latest configuration for bt sip trunks to work with FreePBX?

Nick J

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@leecurt wrote:

I am having problems getting incoming calls working with my sip provider and I’m hoping I could get some help. My provider is https://diamondcard.us/ and their support is really bad so I’m hoping that someone here has an idea of how to get it working. I have outgoing calls working just fine but incoming won’t work. Here is my trunk information,

Outgoing
Trunk Name:
diamondcard

PEER Details:
type=friend
username=XXXXX
fromuser=XXXXX
secret=XXXXXXXXXXXXXXX
host=sip.diamondcard.us
disallow=all
allow=gsm,ulaw
insecure=invite
context=from-diamondcard
fromdomain=sip.diamondcard.us

Incoming
USER Context:
from-diamondcard

USER Details:

Register String:
XXXXX:XXXXXXXXXXXX@sip.diamondcard.us

I’m following their documentation found here
https://wiki.diamondcard.us/podwiki?page=Trixbox

When I try and make an incoming call it rings busy and I get this when I debug asterisk,
[2018-10-08 18:24:38] NOTICE[2267][C-0000000d]: chan_sip.c:26458 handle_request_invite: Call from ‘XXXXX’ (69.65.34.202:5060) to extension ‘s’ rejected because extension not found in context ‘from-diamondcard’.

I’m on a fresh install of the latest FreePBX, I’ve created my extensions and I can call between them and make outgoing calls, I just can’t receive incoming. I assume that I need to add something to the incoming user details but I haven’t figured that out yet. I’ve tried this on 2 different installs so I’m sure that there isn’t a problem with the PBX. I’m going on 3 days now and I know I’m missing something but I can’t tell what it is, please help if you can.

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@pbaeumel wrote:

Dearall,

currently I`m trying to setup a new SIP-Trunk for provider fonial.de.

They have posted general Asterisk-Settings on their page:
https://www.fonial.de/hilfe/trunking/konfiguration-der-telefonanlagen/asterisk/

Register-String: SIP-User:password@sip.solucon.com/SIP-User

For sip.conf the suggest the following settings:
; SIP-User for Registration at fonial-Trunk

[fo279XXXtr24XXXtr_01]
  type=friend
  qualify=yes
  nat=force_rport,comedia
  dtmfmode=RFC2833
  insecure=port,invite
  canreinvite=no
  secret=passwort
  username= fo279XXXtr24XXXtr_01
  fromdomain=sip.solucon.com
  outboundproxy=proxy01.sip.solucon.com
  host=sip.solucon.com
  disallow=all
  allow=ulaw
  allow=alaw 

For the phone they suggest:
; For any phone that is registred on your Asterisk:
[phone]
type=friend
username=phone ; User, set in the phone.
secret=123456 ; Password should be choosen by yourself
host=dynamic
context=dialout ; Change context according to extensions.conf

I have set-up in FreePBX-GUI:

2018-10-08_221944.jpg1881x806 198 KB

2018-10-08_222110.jpg1889x895 203 KB

2018-10-08_222238.jpg1887x892 191 KB

Unfortunately it didn`t work with the settings I have put in the GUI.
Could someone maybe help me converting the general Asterisk-Setting to a FreePBX-Version?

Thanks!

Best regards,
Patrick

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@jdriek wrote:

FreePBX Version 14.0.3.20

I am trying to set the ringtone on inbound calls to a Sangoma phone by setting the Alert Info field to “(Sangoma) Ring 1”. I can see in the asterisk CLI that the Alert-Info is added to the sip header. When I look at the actual sip invite message, the Alert-Info is not being added to the SIP Header. I have tried using the phone configured as both sip and pjsip extensions. No Luck.

PJSIP/305-000006ca Internal Gosub(func-apply-sipheaders,s,1) start
– Executing [s@func-apply-sipheaders:1] NoOp(“PJSIP/305-000006ca”, “Applying SIP Headers to channel”) in new stack
– Executing [s@func-apply-sipheaders:2] Set(“PJSIP/305-000006ca”, “SIPHEADERKEYS=Alert-Info”) in new stack
– Executing [s@func-apply-sipheaders:3] ExecIf(“PJSIP/305-000006ca”, “0?Set(Rheader=1)”) in new stack
– Executing [s@func-apply-sipheaders:4] While(“PJSIP/305-000006ca”, “1”) in new stack
– Executing [s@func-apply-sipheaders:5] Set(“PJSIP/305-000006ca”, “sipheader=ring1”) in new stack
– Executing [s@func-apply-sipheaders:6] ExecIf(“PJSIP/305-000006ca”, “0?Set(Addheader=1)”) in new stack
– Executing [s@func-apply-sipheaders:7] ExecIf(“PJSIP/305-000006ca”, “0?SIPAddHeader(Alert-Info:ring1)”) in new stack
– Executing [s@func-apply-sipheaders:8] ExecIf(“PJSIP/305-000006ca”, “0?Set(PJSIP_HEADER(add,Alert-Info)=ring1)”) in new stack
– Executing [s@func-apply-sipheaders:9] EndWhile(“PJSIP/305-000006ca”, “”) in new stack
– Executing [s@func-apply-sipheaders:4] While(“PJSIP/305-000006ca”, “0”) in new stack
– Executing [s@func-apply-sipheaders:10] ExecIf(“PJSIP/305-000006ca”, “0?SIPRemoveHeader(Alert-Info:)”) in new stack
– Executing [s@func-apply-sipheaders:11] ExecIf(“PJSIP/305-000006ca”, “0?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
– Executing [s@func-apply-sipheaders:12] Return(“PJSIP/305-000006ca”, “”) in new stack
== Spawn extension (from-internal, 305, 1) exited non-zero on ‘PJSIP/305-000006ca’
– PJSIP/305-000006ca Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
– Called PJSIP/305/sip:305@50.75.185.122:51009
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
– Connected line update to PJSIP/Bandwidth_B-000006c9 prevented.
– PJSIP/305-000006ca is ringing
– PJSIP/305-000006ca is ringing

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@superdigi wrote:

i would like to use 2 of 100 extension which is 8000 and 8001 to display random CLID when do the outbound call. below is the current random dial plan with prefix 8 that currently work without permit for certain extension. anyone have any idea to make it work?

[outrt-11] ;
include => outrt-11
exten => _8.,1,Noop
exten => _8.,n,Gosub(pickCallerIDnum,cell${RAND(1,5)},1)
exten => _8.,n,Dial(SIP/trunkname/${EXTEN},30)

[pickCallerIDnum]
exten => cell1,1,Set(CALLERID(num)= 01158628410) ;Specify the numbers that the call should show
same => n,Return
exten => cell2,1,Set(CALLERID(num)= 01157510683);Specify the numbers that the call should show
same => n,Return
exten => cell3,1,Set(CALLERID(num)= 01157515768);Specify the numbers that the call should show
same => n,Return
exten => cell4,1,Set(CALLERID(num)= 01158560794);Specify the numbers that the call should show
same => n,Return
exten => cell5,1,Set(CALLERID(num)= 01158901123);Specify the numbers that the call should show
same => n,Return

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@josephm wrote:

Dear All,

We are using VEGA400 but with the call trace, it is sending double ACK when having incoming call and double 100 trying when doing outside call.

image.png1467x251 52.7 KB

image.png1465x300 49.2 KB

Anyone can help me with this?

Thank you.

Joseph

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@sentinelace wrote:

When I record a parked call, the call is in the control panel but there is no recording file. works fine on a transfer. Is this a bug?

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@faisalkhan wrote:

Hi Guys,

I have to setup freepbx with vodafone in romania as a provider.

it’s a strange setup they have given us a modem with their

SIP CONFIG

Local IP: 192.168.30.2
Local IP CPE intern: 192.168.30.1
Local hostname: ims.vodafone.ro
IP Route: public ip/29 through 192.168.30.1
SIP Gateway primary: public ip
SIP Gateway secondary: public ip

now we have two different networks in the same pbx one with internet and the second with vodafone sip.

I am able to setup the VM with two network adapters but all the traffic is going from one gateway which is for internet not for sip.

how can I setup this with separate gateway for vodafone traffic and separate for internet.

both internet and sip have different subnets.

Anyone can help me with this.

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@robgtelikin wrote:

We have a call that is not in any queue, but is ringing a direct extension, and has been for more than a day. I guess this is a Phantom Call?

Is there a way, either in FOP, command line, or sql query to delete this call?

We prefer not to reboot the server in the middle of the day, and we don’t want to wait until tonight.

Thanks,
Rob

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@itdeptia wrote:

Running FreePBXDistro (not sure which one actually)… so:
Module Admin / Summary:
Current PBX Version: 14.0.3.2
Current System Version: 12.7.4-1712-2.sng7

Reports / Asterisk Info:
Current Asterisk Version: 13.18.5

I have an odd issue on some phones where, every time I “fwconsole reload” some phones display “21 New Voice Mails(s)”. Where the 21 is a random number. And these voicemail boxes are actually empty. Eventually this message just disappears, but comes back next time the reload occurs.

We have Yealink T46S phones in the office.

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@BenBeige wrote:

I’ve got a PBX setup that spans multiple time zones/offices and have defines times zones in voicemail.conf as below:

[zonemessages]

central = America/Chicago|‘vm-received’ Q ‘digits/at’ IMp
pacific = America/Tijuana|‘vm-received’ Q ‘digits/at’ IMp
mountain = America/Denver|‘vm-received’ Q ‘digits/at’ IMp
eastern = America/New_York|‘vm-received’ Q ‘digits/at’ IM

If I set the VM options as:

tz=central

The voicemail envelope randomly reads the word “Twenty” iver and over in the envelope
if it is set

tz=mountain

it inserts the word “Fifth” the same way.

Any ideas as to a cause?

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@Red wrote:

So I have two phone servers with the same public IP but with different private IPs. Phone server A has the vpn server activated using 1194 for remote phones, port forwarding.
On phone server B I want to use softphones using the built in vpn server. Is there anyway to change the vpn port that phone serve B talks across.
I tried to port forward 1194 to server B but it affected server A, since server A also used 1194.
We do not use UCM devices to scan QR codes.
I hope I have explained this for easy understanding.
BTW, I have gotten softphones to work on others phone servers, but only because they only had one phone server on their network, not two. My boss has three virtual phone servers on his network. 1 hosted for remote phones, and two others for internal companies. the vpn is connected between firewalls.

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@mattbratt wrote:

Nice!

So where do I download it, then?

RE:
http://www.digium.com/blog/2018/10/09/asterisk-16-and-freepbx-15-now-available/

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@rramer wrote:

FreePBX 2.3.1.5 Yes I know it’s very very old
and it’s been perking along for a very long time…
recently (as of last week) it has stopped sending the vmail emails.
my counterpart found an error in the logs where there was a 1 inserted in the middle of the FQDN;
he looked at /etc/mail/sendmail.mc, /etc/mail/sendmail.cf, and /etc/asterisk/voicemail.conf; but find no offending form of the address. I logged in to the system today to search further but found nothing.
I performed an nslookup and the FQDN is found.
Does anyone have any idea where the system would getting the fouled address from?
I was able to use the mail cmd and successfully sent myself an email; proving that sendmail isn’t the immediate issue. but I cannot find any other possibility??? a little help please?

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@netphoneusa wrote:

I need to figure out a way for a client to call in to their PBX and record a new Music on Hold file He is not good at logging into his pbx, going to system recordings and doing it that way. He wants to be able to pick up his IP phone and dial a code, enter whatever, and record a new MOH file. Any ideas on this?

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@irish_link wrote:

Hi Everyone!

I have a clean install on a new FirtualboxVM for staging right now. Only thing i have done was update the modules and some of the back end items. yum update style.

Asterisk 13.19.1
FreePBX 14.0.3.19

This was installed from one of the sangoma iso images so I think the OS is SangomaOS. - “SNG7-PBX-64bit-1805-1”

I have a directory server running on a Synology Diskstation that runs LDAP version 3 RFC2251.
I have been able to get other web applications to sync up correctly and have been able to log into said applications with the credentials from Synology Directory Server but no joy with FreePBX under the user management section. I should note that I did not pick Legacy.

I have taken a look at a few forum posts and the “Internal Notes on PBX and OpenLDAP” and they all seem to contradict one another a little bit. The specif way it says to bind is “Bind DN: Must be set to admin LDAP credentials - example: cn=admin,dc=companydnsname,dc=com” however my server shows that the Bind DN should be “uid=root,cn=users,dc=companydnsname,dc=com” If i try to use cn=admin (and have an admin with that specific credentials to use) it does not connect but when I do the way Synology says to bind it will work.

Essentially it “works” but does not sync groups (not a huge deal, I can manage that locally) but even after creating a group and giving access to UCP we can not log in. (giving individual access instead of group doesn’t work either) I keep getting an “Invalid Login Credentials” message. I have tried user@domain.com domain\user domain.com\user and really any other style i could try.

When i click the reset password link, I do get an email with the username as (jsmith) in it so it looks like the username is correct from the UCP stand point.

I should note that I am not experienced when it comes to LDAP. I am currently using it to sync users between different Synology servers and they have a gui that works with each other as expected and a hand holding guide so i have not had to dive into the actual guts of LDAP too much other than the one or two web apps that I got lucky on and it worked right out of the box.

Any suggestions and thoughts would be appreciated. Maybe someone knows a little more about the specific version of LDAP the Synology is running and if it even supports the style that is running on FreePBX.

If it would help i would be happy to post my configuration jsmith@domain.com style.

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@Red wrote:

I don’t know where to start. Our client has been experiencing occasional dropped calls. Their issue is when three calls come in at a time, it drops all three calls. They have given me time stamps and the cdr reports/recorded calls, verify the dropped call, but the logs show no error messages. It just shows the call leaving the simple bridge, first the extension, then the inbound call. I have never dealt with phone systems and network issues till about a year ago. I am still learning, so any expertise advice would be helpful.

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@Matthew99 wrote:

Hi,

I’m setting a queue loop like this:

Queue 1 failovers to Queue 2
Queue 2 failovers to voicemail

calls bounce over to Queue 2 as expected but when not answered in Queue 2 it just rings out is this normal behaviour?

as a test i set Queue 1 falover to Voicemail and works just fine but not when set on the second hop,

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@ELindemann wrote:

Hello @all,

can anybody help me with that?

WARNING[22257] res_odbc.c: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified

WARNING[22257] cdr_adaptive_odbc.c: No such connection ‘asteriskcdrdb’ in the ‘adaptive_connection’ section of cdr_adaptive_odbc.conf. Check res_odbc.conf.

I checked all the files, but always after restarting asterisk this error will not go away. I tried a lot, nothing really helped me to solve this problem.

Other errors while starting asterisk are like:
ERROR[22574] res_sorcery_config.c: Could not create an object of type ‘auth’ with id ‘dcs-trunk-auth’ from configuration file ‘pjsip.conf’
ERROR[22574] res_pjsip/config_auth.c: No authentication username for auth ‘dcs-auth’
But, there, on this part i have no real config @all, therefore this errors are regular.

But how can i solve this
[unixODBC][Driver Manager]Data source …
problem?

I do have this files:
#######################################################
/etc/asterisk/freepbx.conf:

<?php // This file was generated at 2018-06-22T05:12:44+00:00 $amp_conf["AMPDBUSER"] = "freepbxuser"; $amp_conf["AMPDBPASS"] = "password"; $amp_conf["AMPDBHOST"] = "localhost"; $amp_conf["AMPDBNAME"] = "asterisk"; $amp_conf["AMPDBENGINE"] = "mysql"; $amp_conf["datasource"] = ""; require_once "/var/www/html/admin/bootstrap.php"; ?>

#######################################################
/etc/asterisk/odbc.ini

[MySQL-asteriskcdrdb]
Description=MySQL connection to asterisk database
driver=MySQL
server=localhost
database=asteriskcdrdb
username=freepbxuser
password=password
Port=3306
Socket=/var/lib/mysql/mysql.sock
option=3

#######################################################
/etc/odbcinst.ini

[MySQL]
Description = ODBC for MySQL
Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so
Setup = /usr/lib/x86_64-linux-gnu/odbc/libodbcmyS.so
FileUsage = 1

#######################################################
/etc/asterisk/cdr_adaptive_odbc.conf

[adaptive_connection]
connection=asteriskcdrdb
loguniqueid=yes
table=cdr
alias start => calldate

#######################################################
/etc/asterisk/res_odbc.conf

[asteriskcdrdb]
enabled=>yes
dsn=>asteriskcdrdb
pre-connect=>yes
max_connections=>5
username=>freepbxuser
password=>password
database=>asteriskcdrdb

#include res_odbc_custom.conf
#include res_odbc_additional.conf

#######################################################
res_odbc_custom.conf
res_odbc_additional.conf
are blank.
#######################################################

Also i have an asterisk and a freepbxuser in the mysql-DB, you guess, the same password, password.
I am using for all conns/users the same password, it is a testing machine, not productive system, therefore there is no auth-data for connecting to a provider.

But before connecting to my provider, asterisk needs to connect to mysql(?)

*CLI> odbc show status
delivers
ODBC DSN Settings

#####################

*CLI> odbc show
ODBC DSN Settings

Name: asteriskcdrdb
DSN: asteriskcdrdb
Last connection attempt: 2018-10-10 22:03:38
Number of active connections: 0 (out of 5)

######################
Users in mysql:
±-----------------+
| User |
±-----------------+
| asterisk |
| freepbxuser |
| root |
| debian-sys-maint |
| phpmyadmin |
±-----------------+
5 rows in set (0.00 sec)

dbs are:
±-------------------+
| Database |
±-------------------+
| information_schema |
| asterisk |
| asteriskcdrdb |
| mysql |
| performance_schema |
| phpmyadmin |
±-------------------+
6 rows in set (0.00 sec)

Files in /etc/asterisk owns asterisk himself, also other needed dirs too.

Somehow, edting the files above, i lost the overview. Which user has to be connected with which db/dsn and which conf-file will assure with the right configuration the connection.
therefore ONE password.
Later i will change, reinstall with more secure ones.

Why i am getting this errors?

Thx in advance for any appreciated help.
ELindemann

PS:
asterisk-certified-13.21-cert2
complied on debian
3.16.0-4-amd64 #1 SMP Debian 3.16.51-3 (2017-12-13) x86_64 GNU/Linux

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