PBX in a Flash Forum

Dial Plan Help: Prompting people to record their VM greeting...

aaelghat — Fri, 03 Feb 2012 20:45:02 GMT

I know there's a feature where if the extension and VM password are the same, you can have Asterisk guide people through a set-up process, but for various reasons I don't want to use that. I only care about them recording their unavail greeting. Is there some easy way I can have people call into a number, have a prompt for extension and password, and then have them record their greeting without setting the password the same?

I actually do have some code (below), and it works, except if the ID and password don't match it will re-prompt for the password, but won't prompt for the invitation number. So if the invitation number was mistyped, your only choice is to call back.

Any thoughts on the best way of accomplishing what I'd like to do? Thanks in advance.

Code:

[custom-voicemail-newuser]

;this code if for demonstration purposes.

exten => s,1,answer()

exten => s,2,Set(vmnu_dstdir=/var/spool/asterisk/voicemail/default)



;Ask the caller for their mailbox number

exten => s,3,Playback(custom/Welcome-to-LOR)

exten => s,4,read(vmnu_mbx,custom/please-enter-LOR-invitation-number)



;Check the users destination box for existing greeting files.

;Remove any temporary .wav files that might exist if the caller aborted early

exten => s,5,System(rm ${vmnu_dstdir}/${vmnu_mbx}/vmnu_tmpmsg.wav)

exten => s,6,System(ls ${vmnu_dstdir}/${vmnu_mbx}/ | grep wav)



;If there was existing greeting files, this means its an existing

;user and they can go straight to their mailbox (s,8 - s,10)

exten => s,7,GotoIf($["${SYSTEMSTATUS}" = "SUCCESS"]?20:20)

;exten => s,8,Macro(get-vmcontext,${vmnu_mbx})

;exten => s,9,VoiceMailMain(${vmnu_mbx}@${VMCONTEXT})

;exten => s,10,Macro(hangupcall,)



;S,20 verifies the password to the usersmail box

exten => s,20,VMAuthenticate(${vmnu_mbx})



;we tell the user that we are going to step them through a vm process

;exten => s,21,Playback(vm-newuser)

;creates the actual voicemailbox folder

;exten => s,22,System(mkdir ${vmnu_dstdir}/${vmnu_mbx})



;The sections s,23 - s,31 simply set variables for a GOSUB routine.

;It allows the user to be prompted for their NAME, BUSY MESSAGE and UNAVIALABLE MESSAGE

;if you don't want to force the entry of all of them, just change

;the GOSBUB to NoOp and that will bypass the question



exten => s,21,Set(vmnu_msgtype=greet)

exten => s,22,Set(vmnu_msgprmpt=vm-rec-name)

exten => s,23,NoOp(vmnurecord)



exten => s,24,Set(vmnu_msgtype=unavail)

exten => s,25,Set(vmnu_msgprmpt=vm-rec-unv)

exten => s,26,Gosub(vmnurecord)



exten => s,27,Set(vmnu_msgtype=busy)

exten => s,28,Set(vmnu_msgprmpt=vm-rec-busy)

exten => s,29,NoOp(vmnurecord)



;Once all questions are asked, the system puts them into their voicemailbox



;exten => s,30,VoiceMailMain(${vmnu_mbx}@default|s)

exten => s,30,Playback(vm-goodbye)

exten => s,31,Hangup()



;this code prompts and records for each greeting.

exten => s,40(vmnurecord),Playback(${vmnu_msgprmpt})

exten => s,41,Record(${vmnu_dstdir}/${vmnu_mbx}/vmnu_tmpmsg:wav)

exten => s,42,System(chmod +x ${vmnu_dstdir}/${vmnu_mbx}/vmnu_tmpmsg.wav)

;exten => s,43,Background(vm-youhave&recorded)

exten => s,43,System(sox ${vmnu_dstdir}/${vmnu_mbx}/vmnu_tmpmsg.wav ${vmnu_dstdir}/${vmnu_mbx}/vmnu_tmpmsg-copy.wav

exten => s,44,System(sox ${vmnu_dstdir}/${vmnu_mbx}/vmnu_tmpmsg-copy.wav ${vmnu_dstdir}/${vmnu_mbx}/vmnu_tmpmsg.wav compand 0.3\,1 6:-70\,-60\,-20 -5 -90 0.2 vad reverse vad reverse)



exten => s,45,Playback(${vmnu_dstdir}/${vmnu_mbx}/vmnu_tmpmsg)

exten => s,46,Playback(to-accept-recording)

exten => s,47,Playback(1-yes-2-no)

exten => s,48,read(vmnu_yesno||1)

exten => s,49,Gotoif($["${vmnu_yesno}" = "1"]?51:40

exten => s,50,Goto(s,40) ;just in case they manage to go past the gotoif, forces them to re-record

exten => s,51,System(mv ${vmnu_dstdir}/${vmnu_mbx}/vmnu_tmpmsg.wav ${vmnu_dstdir}/${vmnu_mbx}/${vmnu_msgtype}.wav)

exten => s,52,System(cp ${vmnu_dstdir}/${vmnu_mbx}/${vmnu_msgtype}.wav ${vmnu_dstdir}/${vmnu_mbx}/${vmnu_msgtype}.WAV)

exten => s,53,Playback(your-msg-has-been-saved)

;exten => s,54,Playback(auth-thankyou)

exten => s,55,Return

Nmap by Phone

jjolly — Fri, 03 Feb 2012 18:03:31 GMT

A big thank you to all the Asterisk Guru's here for all of the great information here on the forum. I've learned more here than ever could in any Asterisk based book.

I've been pulling my hair out trying to get the following .agi script to work with PIAF but still can't seem to figure out what the problem is. The script allows one to perform an nmap network scan by way of telephone. Any help whatsoever with this will be greatly appreciated.

I am running PIAF Purple + Incredible PBX on CentOS release 5.7 (Final) :32 Bit Kernel: 2.6.18-274.3.1.el5.

I have Perl (v5.15.4) along with the following CPAN modules installed:

Asterisk::AGI
Nmap::Parser
File::Basename
Digest::MD5

I also currently have Cepstral installed along with along with a single Alison 8khz license (installed with the NV script).

My dial plan (extensions_custom.conf) looks like this:

exten => 1337,1,AGI(nmap.agi)
exten => 1337,2,Hangup

I've included the source code below along with another variant that I found.

My CLI output looks like this:

Code:

    -- Executing [1331@from-internal:1] AGI("SIP/501-00000017", "nmap.agi") in new stack

    -- Launched AGI Script /var/lib/asterisk/agi-bin/nmap.agi

    -- Playing 'tts/tts-651d10c4c41504ccd4e3f221e6afeb30' (escape_digits=) (sample_offset 0)

    -- Playing 'tts/tts-818ec742f57be488cb51121f746696b1' (escape_digits=) (sample_offset 0)

    -- AGI Script nmap.agi completed, returning 0

    -- Executing [1331@from-internal:2] Hangup("SIP/501-00000017", "") in new stack

  == Spawn extension (from-internal, 1331, 2) exited non-zero on 'SIP/501-00000017'

    -- Executing [h@from-internal:1] Macro("SIP/501-00000017", "hangupcall") in new stack

    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/501-00000017", "1?skiprg") in new stack

    -- Goto (macro-hangupcall,s,4)

    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/501-00000017", "1?skipblkvm") in new stack

    -- Goto (macro-hangupcall,s,7)

    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/501-00000017", "1?theend") in new stack

    -- Goto (macro-hangupcall,s,9)

    -- Executing [s@macro-hangupcall:9] Hangup("SIP/501-00000017", "") in new stack

  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/501-00000017' in macro 'hangupcall'

  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/501-00000017'

Here's the source for the .agi script that is giving me the CLI log output above:

Code:

#!/usr/bin/perl

#

# AGI Script that prompts the user for an ip address, scans the ip, and reports back to the user.

#

# Requires the Asterisk::AGI and Nmap::Parser perl modules

#



use Asterisk::AGI;

use Nmap::Parser;

use File::Basename;

use Digest::MD5 qw(md5_hex);



sub speak(){

    $text = $_[0];



    my $hash = md5_hex($text);



    my $ttsdir = "/var/lib/asterisk/sounds/tts";

    my $cepoptions = "-p audio/sampling-rate=8000,audio/channels=1";



    my $wavefile = "$ttsdir/tts-$hash.wav";



    unless (-f $wavefile) {



        open(fileOUT, ">/var/lib/asterisk/sounds/tts/say-text-$hash.txt");

        print fileOUT "$text";

        close(fileOUT);



        my $execf="/opt/swift/bin/swift -f $ttsdir/say-text-$hash.txt -o $wavefile $cepoptions";

        system($execf);



        unlink("$ttsdir/say-text-$hash.txt");

    }

    $filename = 'tts/'.basename('tts/'.basename($wavefile,".wav"));

    $AGI->stream_file($filename);

    unlink("$wavefile");



}



$AGI = new Asterisk::AGI;



my %input = $AGI->ReadParse();



my $finished = 0;



&speak("Enter the eye-p address you wish to scan.");



my $ipaddr = '';

my $x = 0;

while (!$finished) {

        my $input = chr($AGI->wait_for_digit('5000'));

        if ($input =~ /^[0-9\*\#]$/) {

                if ($input =~ /^[\*\#]$/) {

                        $x++;

                        if ($x > 3) {

                                $finished = 1;

                        } else {

                                $ipaddr .= '.';

                        }



                } else {

                        $ipaddr .= $input;

                }

        } else {

                        #must have timed out

                        $finished = 1;

        }



        if ( length($ipaddr) > 14) {

                $finished = 1;

        }

}



if ($ipaddr !~ /\d{1,3}\.\d{1,3}\.\d{1,3}\.\d{1,3}/) {

        &speak("Invalid Address: $ipaddr");

        exit 0;

}

&speak("Please hold, now running scan...");



&speak("Please wait");



my $np = new Nmap::Parser;



$nmap_exe = '/usr/bin/nmap';



$np->callback(\&host_handler);



$np->parsescan($nmap_exe,'-sT -p1-1023', $ipaddr);





sub host_handler {

    my $host_obj = shift; #an instance of Nmap::Parser::Host (for current)



        &speak("Host Found with " . $host_obj->tcp_port_count() . " ports open");



    my @ports = $host_obj->tcp_open_ports(); #all ports



        foreach $port (@ports){

                &speak($host_obj->tcp_service($port)->name. " on port" . $port);

        }

        &speak("Scan complete... Thank you...");



        exit;

}



$AGI->exec('Festival', '"No host found"');



exit;

Here is a variant of the same source code displayed above. I only included this for reasons of it possibly being of any help to anyone:

Code:

#!/usr/bin/perl

#

# AGI Script that prompts the user for an ip address, scans the ip, and reports back to the user.

#

# Requires the Asterisk::AGI and Nmap::Parser perl modules

#



use Asterisk::AGI;

use Nmap::Parser;

use File::Basename;

use Digest::MD5 qw(md5_hex);



my $nmap_logfile = "/var/log/asterisk/nmap.log";

my $ttsdir = "/var/lib/asterisk/sounds/tts";



# speaks a string of text

sub speak(){

    $text = $_[0];



    my $hash = md5_hex($text);



    my $wavefile = "$ttsdir/tts-$hash.ulaw";



    unless (-f $wavefile) {



        system("echo $text | text2wave -o $wavefile -otype ulaw -");



    }

    $filename = 'tts/'.basename('tts/'.basename($wavefile,".ulaw"));

    $AGI->stream_file($filename);

    unlink("$wavefile");



}



$AGI = new Asterisk::AGI;



my %input = $AGI->ReadParse();



my ($caller) = $input{callerid} =~ /<(\d+)>/;

if (!defined $caller) {

    ($caller) = $input{callerid} =~ /(\d+)/;

}



my $finished = 0;



&speak("Enter the eye-p address you wish to scan.");



my $ipaddr = '';

my $x = 0;



# While we don't have a complete IP address, have the user enter one

# using '#' for '.'...

while (!$finished) {

        my $input = chr($AGI->wait_for_digit('5000'));

        if ($input =~ /^[0-9\*\#]$/) {

                if ($input =~ /^[\*\#]$/) {

                        $x++;

                        if ($x > 3) {

                                $finished = 1;

                        } else {

                                $ipaddr .= '.';

                        }

                

                } else {

                        $ipaddr .= $input;

                }

        } else {

                        #must have timed out

                        $finished = 1;

        }



        if ( length($ipaddr) > 14) {

                $finished = 1;

        }

}



open(LOG,">>$nmap_logfile");

print LOG localtime(time) . " - $caller - $ipaddr\n";



# Double check the address is valid...

if ($ipaddr !~ /\d{1,3}\.\d{1,3}\.\d{1,3}\.\d{1,3}/) {

        &speak("Invalid Address: $ipaddr");

        exit 0;

}



&speak("Please wait, scan is in progress.");



# Set up a new Nmap::Parser object

my $np = new Nmap::Parser;



$nmap_exe = '/usr/bin/nmap';



$np->callback(\&host_handler);



# Scan the host given.

$np->parsescan($nmap_exe,'-sT -p1-1023', $ipaddr);



# Do this after every host scanned

sub host_handler {

        my $host_obj = shift; #an instance of Nmap::Parser::Host (for current)



        &speak("Host " . $host_obj->hostname() . " found with " . $host_obj->tcp_port_count() . " ports open");



        # Make and array of all the ports.

        my @ports = $host_obj->tcp_open_ports();



        # For every port, speak the port number and service.

        foreach $port (@ports){

                &speak("Open port found, " . $host_obj->tcp_service($port)->name. " on port " . $port);

        }

        &speak("Scan completed! Good bye.");

        

        exit;

        close(LOG);

}



# If a host was found, we shouldn't get here.

&speak("No host found with eye-p address " . $ipaddr . "! Good bye.");



exit;

close(LOG);

Any help in pointing me in the right direction with this will be greatly appreciated.

Regards,
John Jolly

Dell Optiplex 745 $167.99 refurbished

wardmundy — Fri, 03 Feb 2012 15:41:23 GMT

Feb 03, 2012 updates 2062x

darmock — Fri, 03 Feb 2012 15:39:16 GMT

There have been a couple of updates/bug fixes

both red/purple initial install logs were not writing information correctly and this has been corrected in the latest payloads.

There were some errors with our mirrors not having the latest versions of some programs that has been corrected. Primarily it was the whole disk and daily backup programs that were affected. This has been corrected. If you have one of the affected systems simply rerun the install programs for either daily or disk backups.

we have added a new tool which is especially useful for people running vmware

sync-datetime

this will force your time/date to be resynced via ntp immediately Quite handy if you suspend you virtual machine and then restart it a couple of days later.

We have modified the configuration menu for new installs to better explain the freepbx choices that are available to install.

Also we have decided to phase out the daily-backup program in favour of the incredible-backup 2.0 program which is currently undergoing internal beta testing. Once the incredible-backup 2.0 reaches release status the daily-backup program will be removed from distribution. We feel the new Incredible Backup 2.0 is superior in every way and really is a direct replacement. The only feature that will not be in the Incredible Backup 2.0 is the ability to run the Freepbx Backup as part of the daily backup. This can be set up in freepbx directly if you choose to use it.

The other primary reason we are deprecating this program is that the daily-backup program did not have a built in restore program where as the Incredible Backup 2.0 does.

Tom

X-Lite Softphone and Microsoft VPN

peterbata — Fri, 03 Feb 2012 15:24:31 GMT

Hello all.

Most of the working day I will have Microsoft Outlook running. One of my accounts require that I VPN in to a MS Server 2003 SBS located abroad.

As long as the VPN client is running, my X-Lite softphone will not register. I have a feeling it might be that both my laptop and external server are using a similar internal IP range. 192.168.2.x

So far I have had to choose between running X-lite in the background and not checking into one of my email accounts

Your assistance would be greatly appreciated

Travelinman for PIAF2 with CentOS6 install no work - SOLVED

amygrant — Fri, 03 Feb 2012 15:22:33 GMT

Set up a new piaf box and went to install travelinman:

Code:

wget http://incrediblepbx.com/travelinman2.tar.gz

tar zxvf travelinman2.tar.gz

./travelinman2.x

./travelinman2.x: /lib/ld-linux.so.2: bad ELF interpreter: No such file or directory

Solution?

Code:

yum install glibc.i686

then the install works

Code:





                                                                +------------------------System Information---------------------------+

                                                                |  Asterisk   = OFFLINE | Dahdi     = ONLINE  | MySQL     = ONLINE    |

                                                                |  SSH        = ONLINE  | Apache    = ONLINE  | Iptables  = ONLINE    |

                                                                |  Fail2ban   = ONLINE  | Internet  = ONLINE  | Ip6Tables = ONLINE    |

                                                                |  Disk Free  = ADEQUATE| Mem Free  = ADEQUATE| NTPD      = ONLINE    |

                                                                |  SendMail   = ONLINE  | Samba     = OFFLINE | Webmin    = ONLINE    |

                                                                |  Ethernet0  = ONLINE  | Ethernet1 = N/A     | Wlan0     = N/A       |

                                                                |                                                                     |

                                                                |  PIAF Installed Version   = 2.0.6.2 Running on *KVM*                |

                                                                |  FreePBX Version          = 2.9.0.7                                 |

                                                                |  Running Asterisk Version = UNKNOWN                                 |

                                                                |  Asterisk Source Version  = 1.8.8.0                                 |

                                                                |  Dahdi Source Version     = 2.6.0+2.6.0                             |

                                                                |  Libpri Source Version    = 1.4.12                                  |

                                                                |  IP Address               = 192.168.1.20 on eth0                    |

                                                                |  Operating System         = CentOS release 6.2 (Final)              |

                                                                |  Kernel Version           = 2.6.32-220.2.1.el6.x86_64 - 64 Bit      |

                                                                |  Incredible PBX 3 Version = 3.0.3                                   |

                                                                +---------------------------------------------------------------------+

Apple's iCloud Looking Gloomy

wardmundy — Fri, 03 Feb 2012 14:02:41 GMT

Apple iCloud Found to be Infringing on Motorola Patents, German Court Rules

Nice to see Google striking back at the heart of Apple's new business model. In the immortal words of Judge Judy:

Adding Wan interface to live PBX system

jamieautomate — Fri, 03 Feb 2012 11:44:08 GMT

I am just wondering what is the best way to go about adding another network card to be used with a direct WAN feed with its own public IP to a freepbx box.

Currently the freepbx box has one NIC that is on the same lan as the phones and uses the sonicwall to get to the internet.

Can't register more than one extension

truecall — Fri, 03 Feb 2012 02:41:07 GMT

Hello,

I am experiencing a really weird situation. I sure hope someone can help.

I have PIAF installed in a datacenter and it works just great at my home. I have registered 19 phones together placing outbound calls on each one of the phones at the same with NO problems.

Then I go and take the phones to my office so I can ditch our T1, and I can ONLY register 1 phone at the same time. It is the oddest thing.

The difference at the office is we use business class DSL that goes into a clearos server.

Just to eliminate firewall and Intrusion dection on the ClearOS, I added an "ALLOW All" type of rule inbound/outbound on the clearos to the sip server located on the internet. I also disabled the IDS/IPS features within ClearOS.

I hope someone can point me in the right direction. BTW, I have tested this with softphones and grandstream GXP1400's and I am able to duplicate it on the softphones and the Grandstream sip phones.

Many many thanks for the help.

Tim

install-purple.log and suggestion for the installer

Hyksos — Fri, 03 Feb 2012 02:18:16 GMT

Hi,

I'm looking for the install log because I wanted to report some info that is shown during the install as misleading.
I've probably installed piaf around 10 times today because I'm messing with something and while doing that I noted some stuff that maybe could be corrected in the install process.

My concern is that my install-purple.log only contains:

<<>> End install-purple

================================================== =====
install-purple - Finished Log 201202021618
================================================== =====

Maybe that's normal? It contains stuff only when something went wrong? I though it would show most of what flashed on the screen during the install since we're supposed to go look at it?

Also when I install I often go away from the screen, don't you?
When I come back the monitor (and even in vmware) is black.
I usually press some arrow key to bring back display, but if the install is for example at the screen where I have to select the version of freepbx then simply pressing the arrow key counts and the install continues with the default of 2.8.
I just wanted to get the screen back, not input any choice!
I believe other points during the install work the same way.

Am I the only one stressing out during the install? It says I should refrain from touching the keyboard and pressing enter but what am I supposed to do when the screen goes black and simply trying to bring it back on "answers a question of the install" without me wanting to?
Maybe it's me, stupid! Simply enlighten me ;)

Now about the info I think is out of date.

The selection for freepbx has 2.8 has a default and says it's compatible with every known addons.
It says 2.9 won't work reliably with some addons.

It's your choice to default to 2.8, I won't argue with that, your decisions. But saying 2.8 will work with everything and that 2.9 won't? Currently it seems to be the contrary we need to 2.9 for incredible for example, and 2.8 doesn't work.

And we all seem to recommend 2.9 (for addons it's the current standard) but the installer recommend 2.8.

I would like for this to be seen as me trying to help, not me trying to request something or critic a free lunch. I value your work and it's hard to participate actively other than through post like this. I can't modify the file and submit it to save you guys work ;( But i'm sure you have good reason for that and you don't have to answer that.

Thanks

Attached Files

fpbxversions.zip (9.4 KB)

HP ProLiant N40L $299.99

wardmundy — Fri, 03 Feb 2012 00:49:33 GMT

Looks like a good Proxmox server with addition of RAM. Includes 2TB drive at no add'l cost...

Can't login to AvantFAX

freaky al — Thu, 02 Feb 2012 20:55:51 GMT

I saw some threads that mentioned this but I couldn't seem to find an answer. If I missed it, sorry in advance.

I recently installed Incredible Fax 2 on to the latest load of PIAF2 with asterisk 10. I tried to login to the Avantfax admin page with "maint/password" and kept getting error messages about how login failed for wwwadmin. I then used the "Lost your Password?" link and pulled the email with the new temporary password from the sendmail queue. Still no luck. Using phpMyAdmin I can see the password changing for "maint" in the "UserAccount" table of the "avantfax" database, but none of the created passwords work.

Is the error message about the wwwadmin a red herring?

I noticed a solution to a similiar post about Trixbox suggesting that memcached needed to be started, but it doesn't seem to be in use in PIAF. So I don't think that's it.

I saw one solution about how someone rebooted a few times and then it started working. But I don't trust that kind of solution and rebooting a couple times didn't change my luck or fix the problem.

Any thoughts or insights would be much appreciated,
Al

Help needed from registered g+ users

lgaetz — Thu, 02 Feb 2012 19:41:56 GMT

The VUC (Voip Users Conference) will be giving a Kindle Fire away tomorrow during the broadcast to one of 5 people who have previously submitted jokes to be used during the show.

I am in the running!

The voting for the best joke takes place both on google+ and during the broadcast, so I need any registered g+ users to click here:

https://plus.google.com/u/0/10402721...ts/faGvGMT8HxX

scroll down to the very bottom and +1 the joke they like the best. *HINT* you like joke #3 the best.

Thanks for your attention.

Lorne Gaetz

Problem with Sipgate UK

compuguy — Thu, 02 Feb 2012 17:24:26 GMT

I am having a strage problem with Sipgate UK, 2 of my numbers are with them and have been working perfectly for 3.5 years. I am running an old version of PBXIAF which is running FreePbx 2.5.1.5

The problem is that there is no ringing tone being sent to the caller so all they hear is silence. None of my other lines have this problem. All my inward routes go directly to a call queue. Even if I have the queue make an announcement the caller does not hear it. There have been no changes to the system or upgrades in the last 6 months so I am not sure what to look at.

I have tried changing the settings on the trunk as per recommendations by Sipgate.

Does anyone have a working Sipgate configuration that they can post so I can compare to mine?

If I remove the settings in outgoing and the register string the incoming caller hears the ring tone.

Below is the response from Sipgate and below that is my current settings on the trunk:-

Hi Mr. Daly,

Thank you for your e-mail.

A ringing tone is only sent by us to the caller, as long as the call is not answered. Your Asterisk is answering the incoming call automatically as soon as possible to forward the call to the depending partner.
So the ringing is only there in a split second because the call is answered so quick. During the forward to the telephone partner your asterisk does not send out a ringing tone emulation or waiting tone, so the caller just gets silence.

You have to check if you can configure your Asterisk to send some kind of signal to the caller while the call is internally transferred by the Asterisk Unfortunately we cannot assist you on this.

If there is anything else we can help you with, do not hesitate to get in touch with our support again. Thank you!

Best regards,

Trunk Settings:-

username=my username
type=friend
srvlookup=yes
secret=my password
qualify=yes
nat=yes
insecure=very
host=sipgate.co.uk
fromuser=my username
fromdomain=sipgate.co.uk
dtmfmode=rfc2833
disable=all
context=ext-did
canreinvite=yes
authuser=my username
allow=ulaw&alaw

username=my username
type=peer
secret=my password
nat=yes
insecure=very
host=sipgate.co.uk
fromuser=my username
fromdomain=sipgate.co.uk
dtmfmode=rfc2833
disallow=all
context=from-pstn
canreinvite=no
authuser=my username
allow=ulaw&alaw

my username:my password@sipgate.co.uk/my username

Hide caller ID on all outgoing calls

peterbata — Thu, 02 Feb 2012 16:06:37 GMT

I am using a POTS line. If I pickup a handset and wish to hide my caller id (name and number) I simply prefix the outgoing number with *67.

How could I prefix all outgoing calls in this fashion?

I tried entering *67 in Trunk / Outbound Dial Prefix: (call cannot completed)

Outbound Routes / Dial Patterns / Prefix *67 / NXXNXXXXXX (call cannot completed)

Thank you.

Peter

Daily and Disk backup - hang

Fred Rat — Thu, 02 Feb 2012 14:13:53 GMT

Running PIAF 2.0.6.2, purple 32b, FreePBX 2.9.0.9, asterisk 1.8.6.
ran the install-dailybackup and install-diskbackup, edited the two config files in /etc/PBX.

When I run either of the scripts in /etc/cron.daily or /etc/cron.weekly to test, the scripts seem to hang, PS listing below.

I've not found anything useful in /var/log and don't know where else to look.

Please advise on suggestions on either what I did wrong, or where to look for more info, or perhaps even that these scripts aren't working yet on 2.0.6.2.

root@pbx:~ $ ps auxwww | grep back
root 2708 0.0 0.0 1888 296 ? TN Jan30 0:00 /etc/cron.daily/daily-backup.cron
root 2709 0.0 0.0 2520 700 ? SN Jan30 0:00 awk -v progname /etc/cron.daily/daily-backup.cron progname {????? print progname ":\n"????? progname="";???? }???? { print; }
root 2710 0.0 0.0 0 0 ? ZN Jan30 0:00 [daily-backup.cr]
root 5864 0.0 0.0 1868 300 ? TN Feb01 0:00 /etc/cron.weekly/disk-backup.cron
root 5865 0.0 0.0 2520 732 ? SN Feb01 0:00 awk -v progname /etc/cron.weekly/disk-backup.cron progname {????? print progname ":\n"????? progname="";???? }???? { print; }
root 5866 0.0 0.0 0 0 ? ZN Feb01 0:00 [disk-backup.cro]
root 30457 0.0 0.0 4328 732 pts/1 S+ 08:53 0:00 grep back

Piaf on proxmox - latest

theo69 — Thu, 02 Feb 2012 11:40:15 GMT

Hi,

I am running PIAF-2.0.6.1.2-CentOS-6.1-32bit on Proxmox. Bravely/stupidly with Asterisk 10 and freepbx 2.10 All seems to be fine but:

- Fail2ban works, but not for asterisk
- Whitelist script that I have used for a long time appears to install correctly but doesn't block registrations from ip addresses that should not be allowed in?

Fail2ban: it's included so I assume it is meant to work?

Whitelist: is there another script for Centos 6 or should it just work with the existing one? If it should - where would I start looking?

I am running full isos - not the openvz containers.
Thanks!

PIAF as a client do not work but XLite works

kimkhan — Thu, 02 Feb 2012 06:57:41 GMT

Hi,

I have a PIAF-A setup in a remote home.

In my home I have another PIAF-B. I want to connect to the remote PIAF-A. With Xlite I can connect to an extension 123 with no problem at all.

But I want to use my home PIAF-B to register to that same extension 123 that I am able to with Xlite. I have tried many different trunk/peer settings and registratioin strings but my PIAF-B does not registrar.

Xlite only has 3 settings username (Which is extension 123); Password and host IP.

I tried the following settings in my outbound settings:

type=peer
secret=password
username=123
host=remote IP address
fromuser=123
canreinvite=no
insecure=invite,port
qualify=yes
nat=yes

For registration I used:
username:password@IPaddress/username

The reason I want to do this is for the following benefits:

The remote PIAF-A has all my long distance trunks
Local PIAF-B can have multiple extensions and SIP phones connected to the local PIAF-B and then based on dialplan it can dial the local extensions or for long distance the calls will be routed using the remote PIAF-A.

PIAF-A has Freepbx 2.5
PIAF-B was updated to Freepbx 2.9

Anyone here has any idea how to solve this? Thanks in advance for any help.

KK

Asterisk won't start after power problems

freaky al — Thu, 02 Feb 2012 06:39:12 GMT

So my piaf box was working great until we had some power problems with the UPS today. The hardware boots and everything seems to be okay except for asterisk. When I log in to the web interface it has a red error next to asterisk in the server status area. When I ssh in my screen is filled with this ...

Quote:

Asterisk ended with exit status 1
Asterisk died with code 1.
Automatically restarting Asterisk.
Asterisk ended with exit status 1
Asterisk died with code 1.
Automatically restarting Asterisk.

So asterisk is definately not happy. In /var/log/asterisk/full this keeps repeating:

Quote:

[2012-02-01 12:59:18] VERBOSE[14045] config.c: == Parsing '/etc/asterisk/asterisk.conf': [2012-02-01 12:59:18] VERBOSE[14045] config.c: == Found
[Feb 1 12:59:18] VERBOSE[14052] config.c: == Parsing '/etc/asterisk/logger.conf': [Feb 1 12:59:18] VERBOSE[14052] config.c: == Found
[Feb 1 12:59:18] VERBOSE[14052] config.c: == Parsing '/etc/asterisk/logger_general_additional.conf': [Feb 1 12:59:18] VERBOSE[14052] config.c: == Found
[Feb 1 12:59:18] VERBOSE[14052] config.c: == Parsing '/etc/asterisk/logger_general_custom.conf': [Feb 1 12:59:18] VERBOSE[14052] config.c: == Found
[Feb 1 12:59:18] VERBOSE[14052] config.c: == Parsing '/etc/asterisk/logger_logfiles_additional.conf': [Feb 1 12:59:18] VERBOSE[14052] config.c: == Found
[Feb 1 12:59:18] VERBOSE[14052] config.c: == Parsing '/etc/asterisk/logger_logfiles_custom.conf': [Feb 1 12:59:18] VERBOSE[14052] config.c: == Found
[2012-02-01 12:59:18] VERBOSE[14052] config.c: == Parsing '/etc/asterisk/asterisk.conf': [2012-02-01 12:59:18] VERBOSE[14052] config.c: == Found
[2012-02-01 12:59:18] WARNING[14045] db.c: Couldn't prepare statement 'CREATE TABLE IF NOT EXISTS astdb(key VARCHAR(256), value VARCHAR(256), PRIMARY KEY(key))': disk I/O error
[2012-02-01 12:59:18] WARNING[14045] db.c: Couldn't create astdb table: disk I/O error
[2012-02-01 12:59:18] WARNING[14052] db.c: Couldn't prepare statement 'CREATE TABLE IF NOT EXISTS astdb(key VARCHAR(256), value VARCHAR(256), PRIMARY KEY(key))': disk I/O error
[2012-02-01 12:59:18] WARNING[14052] db.c: Couldn't create astdb table: disk I/O error

Disk problems. Okay, so I check the health of the hard disk and get this:

Quote:

root@piaf:/dev $ smartctl -i /dev/sda
smartctl 5.39.1 2010-01-28 r3054 [i686-pc-linux-gnu] (local build)
Copyright (C) 2002-10 by Bruce Allen, http://smartmontools.sourceforge.net

=== START OF INFORMATION SECTION ===
Model Family: Western Digital Scorpio Blue Serial ATA family
Device Model: WDC WD1600BEVT-80A23T0
Serial Number: WD-WXA1A40C5387
Firmware Version: 01.01A01
User Capacity: 160,041,885,696 bytes
Device is: In smartctl database [for details use: -P show]
ATA Version is: 8
ATA Standard is: Exact ATA specification draft version not indicated
Local Time is: Wed Feb 1 12:57:03 2012 EST
SMART support is: Available - device has SMART capability.
SMART support is: Enabled

root@piaf:/dev $ smartctl -s on -d ata /dev/sda
smartctl 5.39.1 2010-01-28 r3054 [i686-pc-linux-gnu] (local build)
Copyright (C) 2002-10 by Bruce Allen, http://smartmontools.sourceforge.net

=== START OF ENABLE/DISABLE COMMANDS SECTION ===
SMART Enabled.

root@piaf:/dev $ smartctl -d ata -H /dev/sda
smartctl 5.39.1 2010-01-28 r3054 [i686-pc-linux-gnu] (local build)
Copyright (C) 2002-10 by Bruce Allen, http://smartmontools.sourceforge.net

=== START OF READ SMART DATA SECTION ===
SMART overall-health self-assessment test result: PASSED

So the hardware is okay, how about the filesystem ...

Quote:

root@piaf:/var/log/asterisk $ fdisk -l

Disk /dev/sda: 160.0 GB, 160041885696 bytes
255 heads, 63 sectors/track, 19457 cylinders
Units = cylinders of 16065 * 512 = 8225280 bytes
Sector size (logical/physical): 512 bytes / 512 bytes
I/O size (minimum/optimal): 512 bytes / 512 bytes
Disk identifier: 0x0007817e

Device Boot Start End Blocks Id System
/dev/sda1 * 1 13 102400 83 Linux
Partition 1 does not end on cylinder boundary.
/dev/sda2 13 19360 155401216 83 Linux
/dev/sda3 19360 19458 785408 82 Linux swap / Solaris

That doesn't look good, but I don't know if that is my problem ... from some research on google it doesn't seem to be an issue.

Quote:

root@piaf:/var/log/asterisk $ sfdisk -uS -l /dev/sda

Disk /dev/sda: 19457 cylinders, 255 heads, 63 sectors/track
Units = sectors of 512 bytes, counting from 0

Device Boot Start End #sectors Id System
/dev/sda1 * 2048 206847 204800 83 Linux
/dev/sda2 206848 311009279 310802432 83 Linux
/dev/sda3 311009280 312580095 1570816 82 Linux swap / Solaris
/dev/sda4 0 - 0 0 Empty

No overlaps. So I booted off a gparted liveCD and and checked all the filesystems (it used e2fs). Everything looked okay, but when I reboot it is still in the same place with asterisk exitting.

This installation was using the PIAF2 CentOS 6.1 build with asterisk 10. I see that a 6.2 version with IncrediblePBX 3 is now available (which I am downloading). Can anyone think of what I should try next, or should I just wipe it and start clean with the latest PIAF2?

Many thanks for taking the time to read through all this.
Al

Google Voice (GV, GVoice) (Incoming calls) being touchy

domiflichi — Thu, 02 Feb 2012 06:04:00 GMT

PIAF Info (from status):
PIAF Installed Version = 2.0.6.1 Running on *HARDWARE*
FreePBX Version = 2.9.0.7
Running Asterisk Version = 1.8.8.0
Asterisk Source Version = 1.8.8.0
Dahdi Source Version = 2.5.0.2+2.5.0.2
Libpri Source Version = 1.4.12
IP Address = 192.168.0.16 on eth0
Operating System = CentOS release 6.2 (Final)
Kernel Version = 2.6.32-131.21.1.el6.i686 - 32 Bit
Incredible Version = 180

Hello,

Let me start by saying I love PIAF! I moved over from trixbox recently (12/26/11) and am not looking back! Thank you everyone for all their hard work for making a great product.

I do have a small problem though. Google Voice (incoming) seems to be finicky. It was working for several weeks as far as I know, but recently I found out that incoming calls were going to Google Voicemail instead of hitting my IVR. When it's working, my IVR picks up after about 1 ring. When it's not working, the caller's phone rings and rings, then gets put into Google Voicemail. Meanwhile, it looks like the calls are hitting my PIAF box because when an incoming call comes in, there's a bunch of activity going on when watching the output of 'asterisk -rvvvvvv'.

After doing some searching I tried adding a 'delay' in my extensions_custom.conf file under a [googlein] heading, which my file didn't even have (a [googlein] heading) so I created one. This didn't seem to help so I removed it. Then I upgraded the FreePBX Google Voice module to 0.6.0, and this didn't seem to help. Then I went to my Google Voice account in FreePBX in the 'Google Voice' section and just clicked 'Submit Changes' (even though i didn't change anything), and then clicked on the Orange 'Apply' button, and then it would work once or twice, then would stop working again.

After more reading, I found out one of the steps in troubleshooting was to have the destination for the Google Voice inbound route go to an extension instead of an IVR. When I did this, incoming calls were successful every time. Then I would change it back to the IVR and it would only work once or twice. I also read something about making sure that port 5222 was allowed in IPTABLES, so I checked and it was there:

Code:

-A INPUT -p udp -m udp --dport 5222 -j ACCEPT

So then I rebooted my server because I noticed that the STATUS screen showed IP6TABLES as offline. This brought IP6TABLES back online and made incoming calls for Google Voice work fine it seemed.

Since I thought it was all working, I finally installed Hamachi VPN and configured it (all using install-hamachi - great script by the way), and then inbound Google Voice calls stopped working again. So I had to disable Hamachi from startup, reboot my server again, and it seems that Inbound Google Voice calls are working again.

I'd like to mention that all this time outbound Google Voice calls work just fine.

I don't understand why I'm having these problems...any ideas or things I can try?
(I would like to have Hamachi VPN running and be able to receive incoming Google Voice calls to my IVR)

Thanks.

incredible PBX 3.0 x64?

tpllc — Thu, 02 Feb 2012 02:29:00 GMT

Im very confused.
on the nerd vittles site the instructions say- Installing PBX in a Flash. Here's a quick tutorial to get PBX in a Flash 2 installed. To use Incredible PBX 3, just install the latest 32-bit or 64-bit version of PBX in a Flash 2. ..

but on soundforge when i click on the 64 bit iso file it says-
NOTE: The 64-bit ISO is NOT compatible with Incredible PBX and requires a DVD because of its size.

which is correct?
i have a dell 2850 with 8 gigs of ram, 2- 3.2Ghz dual core xeons and 6-146GB 10,000 RPM SCSI drives that I would like to run Incredible PBX 3 on.

I'm new to all of this and any help is appreciated.

Wolfram Alpha Setup

thefuzz4 — Wed, 01 Feb 2012 22:41:47 GMT

So I followed all of the instructions to the T from
http://nerdvittles.com/?p=798
Then I dialed 4747 and I hear Elmer tell me to state my question after the beep and press the # key. Immediately after Elmer, Alison comes on and says to disconnect press * otherwise press 1 for english. I press 1 it beeps, I say whatever it is I want to say and then press # and it just sits there. I'm wondering if anyone has any ideas on this or maybe it happened to you too? Thanks for your help with this in advance.

Obi110 star codes & PIAF

The Deacon — Wed, 01 Feb 2012 20:40:40 GMT

Hey guys, is anyone else have a problem using a phone connected to an Obi110 where the star codes don't work on PIAF2?

I've disabled the star codes on the Physical Interfaces / Phone port section of the Obi110 and rebooted a few times, but none of the feature codes seem to work.

I can however get the miscellaneous destinations to work - after an extremely long 15-20 second delay.

Is it just me, or am I completely missing something (obvious)?

System status is here:

Code:

                                                   ┌────────────────────────System Information───────────────────────────┐

                                                   │  Asterisk   = ONLINE  | Dahdi     = ONLINE  | MySQL     = ONLINE    │  

                                                   │  SSH        = ONLINE  | Apache    = ONLINE  | Iptables  = ONLINE    │  

                                                   │  Fail2ban   = ONLINE  | Internet  = ONLINE  | Ip6Tables = ONLINE    │  

                                                   │  Disk Free  = ADEQUATE| Mem Free  = ADEQUATE| NTPD      = ONLINE    │  

                                                   │  SendMail   = ONLINE  | Samba     = OFFLINE | Webmin    = ONLINE    │  

                                                   │  Ethernet0  = ONLINE  | Ethernet1 = N/A     | Wlan0     = N/A       │  

                                                   │                                                                     │  

                                                   │  PIAF Installed Version   = 2.0.6.2 Running on *KVM*                │  

                                                   │  FreePBX Version          = 2.9.0.9                                 │  

                                                   │  Running Asterisk Version = 1.8.8.0                                 │  

                                                   │  Asterisk Source Version  = 1.8.8.0                                 │  

                                                   │  Dahdi Source Version     = 2.6.0+2.6.0                             │  

                                                   │  Libpri Source Version    = 1.4.12                                  │  

                                                   │  IP Address               = 192.168.100.200 on eth0                 │  

                                                   │  Operating System         = CentOS release 6.2 (Final)              │  

                                                   │  Kernel Version           = 2.6.32-220.2.1.el6.i686 - 32 Bit        │  

                                                   │  Incredible PBX 3 Version = 3.0.3                                   │  

                                                   └─────────────────────────────────────────────────────────────────────┘

Immediate PIAF Help NYC & SF

texas1 — Wed, 01 Feb 2012 17:30:22 GMT

We have two offices, one in NYC and one in SF. We're up, but need help with capacity issues, provisioning, etc.

Please respond to me by PM

Forum search feature

peterbata — Wed, 01 Feb 2012 17:19:54 GMT

Hello all. There have been times when I will enter a search phrase for eg. Reload failed because retrieve_conf encountered an error.

The search result message comes back with:

The search term you specified (an) is under the minimum word length (3) and therefore will not be found. Please make this term longer. If this term contains a wildcard, please make this term more specific.

Am I doing something wrong?

Thank you. Peter

Cheap Phones with PA?

FitnessPlus — Wed, 01 Feb 2012 16:52:52 GMT

I'm wanting to upgrade all the phones in my house (5 of them) away from ATA/analog phones to IP based phones. I don't need them to have a bunch of features, but am looking for one particular feature.

I want to be able to use them as a PA, so can say things like, "We're leaving in 5 minutes, get your shoes and socks on" to my 6 children in the house without having to yell.

What phones(s) would you guys recommend I be looking at?

Manager Event CDR not generated

vinod — Wed, 01 Feb 2012 15:21:04 GMT

Hi

PIAF Installed Version = 2.0.6.2
FreePBX Version = 2.8.1.4
Running Asterisk Version = 1.8.8.0

I have enabled the following in cdr_manager.conf
;
; Asterisk Call Management CDR
;
[general]
enabled = yes

I just trying to get the Asterisk Manager CDR after logging in to the AMI through telnet. But could not see any Event: CDR logs generated after making and receiving few test calls.

Please can someone advise what is missing here to enable the Manager CDR.

Your help is much appreciated.

Thanks
Vinod

Digium Launches IP Phones

lgaetz — Wed, 01 Feb 2012 14:26:45 GMT

Looks like more of the same, except:

Quote:

Digium phones include an app engine with a "simple yet powerful JavaScript API" that lets programmers create custom apps that run on the phones.

Note to Digium: Don't do what Aastra did.

http://blog.tmcnet.com/blog/tom-keat...-ip-phones.asp

updates

kenewto — Wed, 01 Feb 2012 03:57:13 GMT

Through no fault of the PIAF software I broke my pbx and had to reload from scratch. After doing so, I find there are some modules available for update.

Is it a good practice to stay current and update them?
Are these updates "Supported"?
Is there any harm in preforming the updates as they come out?

Current config:
PIAF Installed Version = 2.0.6.2 Running on *HARDWARE*
FreePBX Version = 2.9.0.7
Running Asterisk Version = 1.8.8.0
Asterisk Source Version = 1.8.8.0
Dahdi Source Version = 2.6.0+2.6.0
Libpri Source Version = 1.4.12
Operating System = CentOS release 6.2 (Final)
Kernel Version = 2.6.32-220.2.1.el6.x86_64 - 64 Bit
Core 2.9.0.2 Online upgrade available (2.9.0.4)
FreePBX Framework 2.9.0.7 Online upgrade available (2.9.0.9)
Paging and Intercom 2.9.0.5 Online upgrade available (2.9.0.6)

Strange DID routing problem

Cynjut — Wed, 01 Feb 2012 02:41:42 GMT

One of my colleagues is having trouble with his Vitelity account(s). He has two numbers, both route to the same PIAF box (reasonably up-to-date). One of the numbers routes to his phone fine. The other one doesn't - it fails with an 'ss-noservice' message.

If, on the other hand, he includes a "catch-all" inbound route, he can get the call to route to the second line on his phone. The obvious answer is "do that second thing."

The correct answer, though is something else. We've been through the dial-plan with a fine toothed comb.

Here's the output from the CLI and logs.

http://pastebin.com/yiaG4XA9