@yvesc wrote:
i'm using freepbx version 12 with all patch and asterisk 13.6.0. I have an issue with transfert and call in queue. users take call from the queue and do a blind transfert to an extension. for some reason the origine extension that took the call is no longer available until the call is finished on the target extension.
if you think that do not make sense i agree.
YvesC
Posts: 1
Participants: 1
@AndrewZ wrote:
Hello
Just realized that DID column in CDR is empty. Reinstalling CDR module 12.0.23 gives no change.
There is a note in a changelog '12.0.21: Add index to DID col in cdr DB' which means for me that the similar issue has been discovered and possibly solved... Any idea how can I get back the DID information?
BTW, still see the old records with DID information from November '15 and earlier.
Thanks!
Posts: 1
Participants: 1
@OttisC wrote:
I first want to apologize if I don't have this in the correct category Our company is currently running on Trixbox and I am in the process of creating our new FreePBX system. Unfortunately, I was not part of the initial setup of our Trixbox system but I have the sole responsibility of upgrading to FreePBX. My question resides on how to address external users for connection to the FreePBX server. Our current process is through opening ports using the iptables however this can be fairly intensive even for smaller company due to home users getting a new IP address. Our phones currently connection using FTP and I understand that FreePbx default method is through TFTP. I am looking for some design suggestions for a more efficient way of supporting users not on our network while still maintaining an acceptable level of security above the secret. Thanks again for your time.
Posts: 1
Participants: 1
@asyscom wrote:
Hello to all , I will first describe the situation .
FreePBX it is connected to a Patton sn4120 with 2 BRI , 4 SIP phones connected to the PBX and 22 analog phones connected to the PBX via a grandstream gxw4200 fxs gateway .
The calls come at a Ringroup doing play 7 phones simultaneously ( 3 sip and 4 analog and 1 cordless ) .
That's it in terms of configuration and it works fine . What the customer asks me ( with the old analog exchange that I replaced it worked) is diversifying the sound of the 6 internal Ringroup depending on whether the call comes from outside or inside .
The models of phones are
3 Grandstream gxp1610
3 ST - 200 ( analog )
1 Brondi Cruise ( cordless )
I do not know why but it seems that this feature to the customer is vitalThanks in advance
Posts: 1
Participants: 1
@Propaganda wrote:
Hi,
Has anyone successfully setup a trunk to Draytel.org? Incoming calls work but outgoing calls fail. Asterisk attempts to route the outgoing call but then fails and 'All circuits are busy now' is played.
Any help or pointers would be very gratefully received. This is driving me mad.
FreePBX 12.0.76.2
Here is my trunk config:
Trunk name: Draytel
Peer details:
type=friend
username=8*****
secret=5*****
fromuser=8*****
host=draytel.org
dtmfmode=rfc2833
fromdomain=draytel.org
context=from-pstn
insecure=very
qualify=yes
disallow=all
allow=ulaw'sip show registry' outputs:
Host dnsmgr Username Refresh State Reg.Time
draytel.org:5060 N 8***** 105 Registered Fri, 15 Jan 2016 20:55:59
1 SIP registrations.Here is the output of sip 'set debug ip draytel.org' when I place an outgoing call:
SIP Debugging Enabled for IP: 217.14.138.127
Reliably Transmitting (NAT) to 217.14.138.127:5060:
OPTIONS sip:draytel.org SIP/2.0
Via: SIP/2.0/TCP 81.142.231.116:5060;branch=z9hG4bK7e7c9184;rport
Max-Forwards: 70
From: "Unknown" sip:8*****@netphone.domain.co.uk;tag=as433bdce6
To:
Contact: @81.142.231.116:5060;transport=TCP>
Call-ID: 6e21940010615d0c5983d4221d6124f7@netphone.domain.co.uk
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.2(12.8.2)
Date: Fri, 15 Jan 2016 20:57:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [0*********2@from-internal:1] Macro("SIP/6003-0000000c", "user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/6003-0000000c", "TOUCH_MONITOR=1452891481.60") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/6003-0000000c", "AMPUSER=6003") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/6003-0000000c", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/6003-0000000c", "1?Set(REALCALLERIDNUM=6003)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/6003-0000000c", "AMPUSER=6003") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/6003-0000000c", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/6003-0000000c", "AMPUSERCIDNAME=John Smith") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/6003-0000000c", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/6003-0000000c", "AMPUSERCID=6003") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/6003-0000000c", "_DIALOPTIONS=Ttr") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/6003-0000000c", "CALLERID(all)="John Smith" <6003>") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/6003-0000000c", "0?limit") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("SIP/6003-0000000c", "1?Set(GROUP(concurrency_limit)=6003)") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/6003-0000000c", "1?continue") in new stack
-- Goto (macro-user-callerid,s,27)
-- Executing [s@macro-user-callerid:27] Set("SIP/6003-0000000c", "CALLERID(number)=6003") in new stack
-- Executing [s@macro-user-callerid:28] Set("SIP/6003-0000000c", "CALLERID(name)=John Smith") in new stack
-- Executing [s@macro-user-callerid:29] Set("SIP/6003-0000000c", "CDR(cnum)=6003") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/6003-0000000c", "CDR(cnam)=John Smith") in new stack
-- Executing [s@macro-user-callerid:31] Set("SIP/6003-0000000c", "CHANNEL(language)=en") in new stack
-- Executing [0*********2@from-internal:2] Gosub("SIP/6003-0000000c", "sub-record-check,s,1(out,0*********2,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/6003-0000000c", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("SIP/6003-0000000c", "_RECSTATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("SIP/6003-0000000c", "NOW=1452891481") in new stack
-- Executing [s@sub-record-check:4] Set("SIP/6003-0000000c", "__DAY=15") in new stack
-- Executing [s@sub-record-check:5] Set("SIP/6003-0000000c", "__MONTH=01") in new stack
-- Executing [s@sub-record-check:6] Set("SIP/6003-0000000c", "__YEAR=2016") in new stack
-- Executing [s@sub-record-check:7] Set("SIP/6003-0000000c", "__TIMESTR=20160115-205801") in new stack
-- Executing [s@sub-record-check:8] Set("SIP/6003-0000000c", "__FROMEXTEN=6003") in new stack
-- Executing [s@sub-record-check:9] Set("SIP/6003-0000000c", "_MONFMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("SIP/6003-0000000c", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("SIP/6003-0000000c", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("SIP/6003-0000000c", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("SIP/6003-0000000c", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("SIP/6003-0000000c", "3?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("SIP/6003-0000000c", "1?sub-record-check,out,1") in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] NoOp("SIP/6003-0000000c", "Outbound Recording Check from 6003 to 0*********2") in new stack
-- Executing [out@sub-record-check:2] Set("SIP/6003-0000000c", "RECMODE=dontcare") in new stack
-- Executing [out@sub-record-check:3] ExecIf("SIP/6003-0000000c", "1?Goto(routewins)") in new stack
-- Goto (sub-record-check,out,7)
-- Executing [out@sub-record-check:7] Gosub("SIP/6003-0000000c", "recordcheck,1(dontcare,out,0*********2)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("SIP/6003-0000000c", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("SIP/6003-0000000c", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("SIP/6003-0000000c", "") in new stack
-- Executing [out@sub-record-check:8] Return("SIP/6003-0000000c", "") in new stack
-- Executing [0*********2@from-internal:3] Set("SIP/6003-0000000c", "MOHCLASS=default") in new stack
-- Executing [0*********2@from-internal:4] Set("SIP/6003-0000000c", "_NODEST=") in new stack
-- Executing [0*********2@from-internal:5] Macro("SIP/6003-0000000c", "dialout-trunk,1,0*********2,,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/6003-0000000c", "DIAL_TRUNK=1") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/6003-0000000c", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/6003-0000000c", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/6003-0000000c", "DIAL_NUMBER=0*********2") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/6003-0000000c", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/6003-0000000c", "OUTBOUND_GROUP=OUT_1") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/6003-0000000c", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/6003-0000000c", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/6003-0000000c", "DIAL_TRUNK_OPTIONS=Tt") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/6003-0000000c", "outbound-callerid,1") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/6003-0000000c", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/6003-0000000c", "0?Set(REALCALLERIDNUM=6003)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/6003-0000000c", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/6003-0000000c", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/6003-0000000c", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/6003-0000000c", "TRUNKOUTCID=<0*********5>") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/6003-0000000c", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,14)
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/6003-0000000c", "1?Set(CALLERID(all)=<0*********5>)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/6003-0000000c", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/6003-0000000c", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/6003-0000000c", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:18] Set("SIP/6003-0000000c", "CDR(outbound_cnum)=0*********5") in new stack
-- Executing [s@macro-outbound-callerid:19] Set("SIP/6003-0000000c", "CDR(outbound_cnam)=") in new stack
[2016-01-15 20:58:01] WARNING[2136]: func_cdr.c:349 cdr_write_callback: CDR requires a value (CDR(variable)=value)
)� -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/6003-0000000c", "0?sub-flp-1,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/6003-0000000c", "OUTNUM=0*********2") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/6003-0000000c", "custom=SIP/Draytel") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/6003-0000000c", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/6003-0000000c", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/6003-0000000c", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/6003-0000000c", "") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/6003-0000000c", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/6003-0000000c", "1?Set(CONNECTEDLINE(num,i)=0*********2)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/6003-0000000c", "1?Set(CONNECTEDLINE(name,i)=CID:0*********5)") in new stack
-- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/6003-0000000c", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:22] Dial("SIP/6003-0000000c", "SIP/Draytel/0*********2,300,Tt") in new stack
[2016-01-15 20:58:01] WARNING[10468][C-0000000b]: app_dial.c:2421 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:23] NoOp("SIP/6003-0000000c", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20") in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/6003-0000000c", "0?continue,1:s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/6003-0000000c", "RC=20") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/6003-0000000c", "20,1") in new stack
-- Goto (macro-dialout-trunk,20,1)
-- Executing [20@macro-dialout-trunk:1] Goto("SIP/6003-0000000c", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/6003-0000000c", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 20 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] Set("SIP/6003-0000000c", "CALLERID(number)=6003") in new stack
-- Executing [0*********2@from-internal:6] Macro("SIP/6003-0000000c", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/6003-0000000c", "") in new stack
-- Executing [s@macro-outisbusy:2] Playback("SIP/6003-0000000c", "all-circuits-busy-now,noanswer") in new stack
-- Playing 'all-circuits-busy-now.ulaw' (language 'en')0x7f16450ef3a0 -- Probation passed - setting RTP source address to 192.168.1.204:5004
-- Executing [s@macro-outisbusy:3] Playback("SIP/6003-0000000c", "pls-try-call-later,noanswer") in new stack
-- Playing 'pls-try-call-later.ulaw' (language 'en')
Really destroying SIP dialog '6e21940010615d0c5983d4221d6124f7@netphone.domain.co.uk' Method: OPTIONS
[2016-01-15 20:58:04] WARNING[10468][C-0000000b]: app_playback.c:493 playback_exec: Playback failed on SIP/6003-0000000c for pls-try-call-later,noanswer
-- Executing [h@from-internal:1] Hangup("SIP/6003-0000000c", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/6003-0000000c'
Reliably Transmitting (NAT) to 217.14.138.127:5060:
OPTIONS sip:draytel.org SIP/2.0
Via: SIP/2.0/TCP 81.142.231.116:5060;branch=z9hG4bK5fe1f205;rport
Max-Forwards: 70
From: "Unknown" sip:8*****@netphone.domain.co.uk;tag=as64a29b4e
To:
Contact: @81.142.231.116:5060;transport=TCP>
Call-ID: 70935a060e28312d261d78147ebf1070@netphone.domain.co.uk
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.76.2(12.8.2)
Date: Fri, 15 Jan 2016 20:58:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Posts: 1
Participants: 1
@bigunk wrote:
I have a request prior to deploying a FreePBX for a client. In this case, it's a doctor, and he has 2 published numbers. One for general office, and one for his surgery center. He wants to be able to output either number as he makes an outbound call. Doing it on his analog lines is simple. Just select the line and dial. Now I know we can assign an outbound ANI per phone on the PBX, but this client wants to do it from each phone by pressing a button prior to dialing.I can do it with other PBX's, Panasonic for example. But can I do this on a FreePBX using Poly or Panasonic phones, or something like them?
I call this ethical spoofing because this guy has the numbers already, and just wants to use them outbound.
Posts: 4
Participants: 4
@claloano wrote:
.htaccess file
I need to edit the .htaccess file in the dir freePBX
directives are about as
RewriteEngine on
RewriteCond% {REMOTE_ADDR}! ********
RewriteCond% {REMOTE_ADDR}! ^ 127 \ .0 \ .0 \ .1
RewriteCond% {REQUEST_URI}! /........./$ [CN]
RewriteRule. * / Ergotel / [R = 301, L]The fact that the restart or more likely to find the upgrade freePBX with the original settings
how should I do?
Posts: 1
Participants: 1
@claloano wrote:
I am preparing a FreePBX
for the moment it has no active trunk to make and receive calls
the interiors are registered but
I received this message, and I can not understand what it means ...
Device Dora (49) had an invalid emergency callerid and was set to blank
Vito that is marked as a critic I might ask support
Posts: 1
Participants: 1
@pasco wrote:
Hi
I recently upgraded my freepbx distro from 12 to 13. Everything worked out sweet as I supposed so far. But now I noticed that UCP doesn't show Voicemail at all anymore. The messages are still at the right places on the server I guess, wav- & txt.files. But in the UCP there is no option to choose "voicemail".
Any hints on how fixing that?
Thanks & cheers
p@sco
Posts: 1
Participants: 1
@steveh42 wrote:
After upgrading to FreePBX 13, users cannot login to UCP.
I have tries creating a test user in User Manager and setting the UCP login permission to 'Yes' but the test user cannot login.
The ucp_err.log in /var/log/asterisk has a message that states "There was an error with MySQL Connection".
Does anyone have any ideas where to look next?
Thanks.
Posts: 1
Participants: 1
@siv002 wrote:
Hello.
Installed Asterisk and FreePBX 12 11. Installed Gateway DVG-7022s.
In the FreePBX created extension 2341800 with the value of host = 192.168.3.186. Outgoing calls to port pass. Incoming calls to the port does not pass. The problem is solved by briefly the change in the value of host to dynamic and
registration port. After such registration, and change in the host 192.168.3.186 calls pass. Before restarting Asterisk.
On the page "Chain_Sip Info" it is clear that the expansion of value appears username.
This problem can not be observed from the trunks.
How do I fix this situation? So to be able to call on the FXS port gateway, which assigned extension with a static host.
Sincerely.
Igor Stepanenko.
Posts: 1
Participants: 1
@paulkasper wrote:
Hi all,
I've been using the new Bulk Handler to import extensions and DIDs into a new system, and have not been able to successfully update max_contacts. I see two sections in my csv that include max contacts, specifically:
max_contacts, which is located between match and maximum_expiration
max_contact, which is located between mailbox and media_use_received_transport.I've set a single Max Contact in Application -> Extensions -> (choose extension) -> Advanced -> Max Contacts (located between Account Code and Media Use Received Transport), which shows up under max_contact for an export of extensions in the Bulk Handler. I can then copy this value to other extensions, but it never sticks. It even reverts the single extension back to 1 max contact when re-importing the csv back into Bulk Handler.
I've also tried setting max_contacts to the same values, but this doesn't appear to ever stay written.
Is there something I'm missing?
Posts: 1
Participants: 1
@ryanhoutz wrote:
I have a FreePBX server Asterisk 13 running in our location and we have a remote office connected via VPN. We are both on a 10.240.???.??? LAN but when calling an extension in the remote office the server tells the client to redirect the traffic to the WAN IP. The calls do go through but I don't understand why the server would be redirecting the traffic. Anyone else noticed this? Do I have something configured incorrectly? Any help or advise would be a big help.
Posts: 2
Participants: 2
@venice89 wrote:
Hi,
I am playing around with FreePBX. Actually I have an Cisco Callmanager and I plan to change vom Cisco to FreePBX.
A nice and very Important feature is the PickUp group.Example: A Person is Calling the 231 and just this phone is ringing, but on all phones inside this Pickup group I can see who is calling and I can Pick-up the call from every phone inside the group.
Is this possible with FreePBX? Or is it depending on the SIP Phone?Greetings
Tobias
Posts: 1
Participants: 1
@lhoward96 wrote:
I've been trying to install FreePBX using a Linode StackScript, however I keep running into the prompts that come up when I use
./install_amp --installdb. I'm using FreePBX 12 and CentOS 6.5 for the installation. I basically copied the install commands from the wiki into a script.The problem I'm running into is that when
./install_amp --installdbcomes around in this script, the prompts for username, password, and all the default options pop up. Is there a way to make this more automated? Like change the amportal.conf before it gets to that part? Or send a simulated 'enter'?#!/bin/bash # Disable selinux and update system sed -i 's/\(^SELINUX=\).*/\SELINUX=disabled/' /etc/sysconfig/selinux yum -y update yum -y groupinstall core yum -y groupinstall base # Install required dependencies yum -y install gcc gcc-c++ lynx bison mysql-devel mysql-server php php-mysql php-pear php-mbstring tftp-server httpd make ncurses-devel libtermcap-devel sendmail sendmail-cf caching-nameserver sox newt-devel libxml2-devel libtiff-devel audiofile-devel gtk2-devel subversion kernel-devel git subversion kernel-devel php-process crontabs cronie cronie-anacron wget vim php-xml uuid-devel libtool sqlite-devel # Turn off iptables and turn on mysql/httpd chkconfig --level 0123456 iptables off service iptables stop chkconfig --level 345 mysqld on service mysqld start chkconfig --level 345 httpd on service httpd start # Install PearDB pear channel-update pear.php.net pear install db-1.7.14 # Google voice dependencies cd /usr/src wget https://iksemel.googlecode.com/files/iksemel-1.4.tar.gz tar xf iksemel-*.tar.gz cd iksemel-* ./configure make make install # Add asterisk user adduser asterisk -M -c "Asterisk User" # Download asterisk source files cd /usr/src wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz wget http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4-current.tar.gz wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz git clone https://github.com/akheron/jansson.git wget http://www.pjsip.org/release/2.2.1/pjproject-2.2.1.tar.bz2 # Compile and install pjproject cd /usr/src tar -xjvf pjproject-2.2.1.tar.bz2 cd pjproject-2.2.1 CFLAGS='-DPJ_HAS_IPV6=1' ./configure --prefix=/usr --enable-shared --disable-sound\ --disable-resample --disable-video --disable-opencore-amr --libdir=/usr/lib64 make dep make make install # Compile and Install jansson cd /usr/src/jansson autoreconf -i ./configure --libdir=/usr/lib64 make make install # Compile and install Asterisk cd /usr/src tar xvfz asterisk-13-current.tar.gz rm -f asterisk-13-current.tar.gz cd asterisk-* contrib/scripts/install_prereq install ./configure --libdir=/usr/lib64 contrib/scripts/get_mp3_source.sh make make install make config ldconfig # Install asterisk-extra-sounds mkdir -p /var/lib/asterisk/sounds cd /var/lib/asterisk/sounds wget http://downloads.asterisk.org/pub/telephony/sounds/asterisk-extra-sounds-en-wav-current.tar.gz tar xfz asterisk-extra-sounds-en-wav-current.tar.gz rm -f asterisk-extra-sounds-en-wav-current.tar.gz wget http://downloads.asterisk.org/pub/telephony/sounds/asterisk-extra-sounds-en-g722-current.tar.gz tar xfz asterisk-extra-sounds-en-g722-current.tar.gz rm -f asterisk-extra-sounds-en-g722-current.tar.gz # Install and configure FreePBX cd /usr/src wget http://mirror.freepbx.org/modules/packages/freepbx/freepbx-12.0-latest.tgz tar vxfz freepbx-12.0-latest.tgz # Change ownership chown asterisk. /var/run/asterisk chown -R asterisk. /etc/asterisk chown -R asterisk. /var/{lib,log,spool}/asterisk mkdir /usr/lib/asterisk chown -R asterisk. /usr/lib/asterisk chown -R asterisk. /usr/lib64/asterisk mkdir /var/www/html chown -R asterisk. /var/www/ # Modifications to apache sed -i 's/\(^upload_max_filesize = \).*/\120M/' /etc/php.ini cp /etc/httpd/conf/httpd.conf /etc/httpd/conf/httpd.conf_orig sed -i 's/^\(User\|Group\).*/\1 asterisk/' /etc/httpd/conf/httpd.conf service httpd restart #Configure asterisk db cd /usr/src/freepbx export ASTERISK_DB_PW=amp109 mysqladmin -u root create asterisk mysqladmin -u root create asteriskcdrdb mysql -u root -e "GRANT ALL PRIVILEGES ON asterisk.* TO asterisk@localhost IDENTIFIED BY '*********';" mysql -u root -e "GRANT ALL PRIVILEGES ON asteriskcdrdb.* TO asterisk@localhost IDENTIFIED BY '********';" mysql -u root -e "flush privileges;" #Restart asterisk and install FreePBX cd /usr/src/freepbx ./start_asterisk start ./install_amp --installdb --username=asterisk --password=******* amportal chown amportal a ma installall amportal a reload amportal a ma refreshsignatures amportal chown #Start FreePBX ln -s /var/lib/asterisk/moh /var/lib/asterisk/mohmp3 amportal restart # Install and setup commercial modules wget -P /etc/yum.repos.d/ -N http://yum.schmoozecom.net/schmooze-commercial/schmooze-commercial.repo yum clean all yum -y install php-5.3-zend-guard-loader sysadmin fail2ban incron ImageMagick # Restart Apache and install sysadmin service httpd restart amportal a ma download sysadmin amportal a ma install sysadmin sed -i '338d' /etc/httpd/conf/httpd.conf sed -i '338i AllowOveride ALL' /etc/httpd/conf/httpd.conf amportal chown amportal a ma refreshsignatures amportal a reload
Posts: 1
Participants: 1
@remosti wrote:
Hello Everybody
I have a problem with the UCP. I have set the User Manager to get the Users over AD from my Domain-Controller, whitch works. I also set the access-rights to the UCP for the User to true. But i cant login to the UCP.
No matter what i input as Username (username as it shows in UserManager, domain\username, username@domain) i always get an "Invalid Login Credentials"-Error back. I searched in the Internet, but couldn't find anything.Does anybody uses the UCP with AD-Authentication? What is the correct Username-Style? Are there some settings, whitch need to be adjusted?
Thank you for your help
Remo Stirnimann
Posts: 2
Participants: 2
@parsam wrote:
I have installed FreePBX 12.0.76.2. My extensions registered to server via FreePBX UCP.
Usually everything works fine, but sometimes my FreePBX web interface becomes unavailable, extensions can't register to FreePBX and I can't connect to FreePBX admin page. When I connect to server through ssh I see high CPU load. To resolve the issue I restart asterisk service then apache service and then kill /var/www/html/admin/modules/dashboard/scheduler.php process.Please help me to find cause of problem,
Thank you.
Posts: 5
Participants: 2
@GRZMRC wrote:
Hi,
Yesterday I upgraded my freepbx12 to freepbx13.
In usermanager I enabled the fax for a user
In inbound route I enabled the fax detection (SIP) and in fax destination I selected the extension of a fax enabled user.Now, when someone send me a fax, I ear the extension phone ringing.
Can you help me?
In freepbx12 the fax works greatPS: in fax configuration I read FAx drivers supported by this module are: Fax for asterisk (res_fax_digium) with license/Spandsp based app_fax (res_fax_spandsp)
Posts: 2
Participants: 1
@tennet wrote:
After changing my voip Gateway last night and moving it to MP104 i found that incomming calls are not reaching my voip gateway.
After digging in my pbx logs and some debuging on the sip i found that the pbx for incomming calls was passing the header Alert-Info.
The Alert-Info in the inbound route was set to none so this header should not have been present.
Digging further i found that in the databse in table incomming the alertinfo field is set to: ' ' and not '' it contains a space. This space makes the macro-dial-one introduce the Alert-Info header in the sip headers.
How can i get in touch with some one from freepbx to modify the core in order to take into consideration this fact.
I suggest a change from:
exten => s,n,GosubIf($["${ALERT_INFO}"!=""]?func-set-sipheader,s,1(Alert-Info,${ALERT_INFO}))
to:
exten => s,n,GosubIf($["${ALERT_INFO}"!="" && ["${ALERT_INFO}"!=" "]?func-set-sipheader,s,1(Alert-Info,${ALERT_INFO}))So it takes into account the space. I also tried to modify the web page to trim the space before updateting the db table but it makes the Alert Info select not work correctly.
Posts: 6
Participants: 2
@cblack wrote:
Hello. This is my first post, so I hope I have this in the correct category.
We are having issues with a SIP extension on a WAN losing the connection to the FreePBX server. It happens on internal extension calls and outgoing calls. (LAN extensions work great.)
We just newly installed FreePBX 13, but we have had this problem on an older version first. We have also tried the PBX server and SIP extension at different locations with different routers... still happens.
Currently, we run the following:
Asterisk 13.5.0
FreePBX 13.0.51The SIP extension phone is a Panasonic KX-TGP500B04.
The problems are as follows:
- Sometimes the connection just drops.
- Sometimes one direction stops working, but the other can hear fine.
- Sometimes the connection becomes very choppy. The voice sounds mechanized... then eventually drops.The only common denominators I can see are the SIP settings in the Panasonic phone on the WAN and the fact we are using default settings for SIP configuration in FreePBX.
Does anyone have any suggestions of what configuration settings to modify?
Thank you.
Posts: 1
Participants: 1
