Add on RPM's

Posted April 14th, 2011 by tonyclewis (Developer)

I wanted to inform everyone of some 3rd party RPM's we have made avaialble in our repos. Luke is hard at work in adding a simple RPM Magament Section to the Sysadmin Module that will let you enable 3rd Party RPM's from the GUI but for the time being you can yum install any of the following RPM's safely.

fop2-freepbx - this will replace fop1 on the system with fop2 and make the link on the landing page point to the new fop2. It will also install the fop2 FreePBX style modules to manage fop2.

LumenVox* - This will install all the RPM's and Connector bridges for LumenVox Speech Recg to work. Make sure to include the * in the name as their are numerous packages that LumenVox needs to install.

aastra-xml-scripts - This will install all the great Aastra XML scripts that everyone has come to love.

aastra-ipphone - This will install the Aastra Firmware Pack for the 67 series phones.

wanpipe - - This will install the wanpipe drivers for Sangoma PSTN cards. Once installed you can type setup-sangoma to have the system take you through a wizard to setup your Sangoma Card.

If anyone has any other RPM's they would like to see added please let me know by replying to this thread.

Thanks

__________________

Tony Lewis
Schmooze Com, Inc.
FreePBX Developer

Faxing Solution

On April 15th, 2011 depster (tadpole) said:

Would like to see a rpm/script written for a faxing solution. Perhaps AvantFax & Hylafax.

New: Incredible Fax

On April 28th, 2011 wardmundy (Contributor) said:

We have a new (experimental) Avantfax/Hylafax installer for CentOS 5.6, PIAF-Purple with Asterisk 1.8 and FreePBX 2.8, and Incredible PBX that you are most welcome to try. Here's the link: http://pbxinaflash.com/forum/showthread.php?p=64291#post64291

With the Fax Configuration

On April 15th, 2011 tonyclewis (Developer) said:

With the Fax Configuration module included in FreePBX that we built over a year ago and Fax For Asterisk (res_fax_digium) you get flawless inbound faxing. We are working on getting the res_fax_Spandsp built into the distro as a free alternative to buying the Fax For Asterisk.

I am not a fan of Hylafax and have just seen to many rogue process issues with it and than the control of setting up Hylafax and IAX Modems is tough. We try to let everyone control everything from the GUI and not drop down to the command line for anything if we can help it.

We do have a 3rd party commercial add on coming out shortly that will allow outbound faxing from the ARI and also view all inbound faxes from the ARI. It also lets you setup coversheet with your logo and everything included and control the settings of the coversheet on a per user basis from the ARI. Its a module we have been using for years on our commercial side.

__________________

Tony Lewis
Schmooze Com, Inc.
FreePBX Developer

OK we just built new 1.8.3

On April 15th, 2011 tonyclewis (Developer) said:

OK we just built new 1.8.3 RPM's with spandsp configured. We have also built a spandsp RPM. I will get the distro updated to include the new asterisk RPM's and spandsp and build a upgrade script for any users that have the current distro installed to test.

This should all work with the Fax Configuration module in FreePBX to give you free inbound faxing using res_fax and res_fax_spandsp.

Is there anyone out there who can test this for us?

__________________

Tony Lewis
Schmooze Com, Inc.
FreePBX Developer

OK the distro has been

On April 15th, 2011 tonyclewis (Developer) said:

OK the distro has been updated to include res_fax_spandsp and the spandsp library for asterisk. Anyone installing it from this point forward will get these changes. The version that is including these changes is the 1.8.1.2-3.

Follow this thread on how to upgrade existing systems to the latest version

http://www.freepbx.org/forum/freepbx-distro/distro-discussion-help/relea...

__________________

Tony Lewis
Schmooze Com, Inc.
FreePBX Developer

I applied the upgrade but I

On April 15th, 2011 dcitelecom (tadpole) said:

I applied the upgrade but I don't think it's working.
The new version info is 1.8.1.2-3
I did an amportal restart and even rebooted the system

When I select fax on inbound route the only 2 choices are Dahdi and SIP. Nvfax is there but it's greyed out.

Are you faxing in from a Sip

On April 15th, 2011 tonyclewis (Developer) said:

Are you faxing in from a Sip trunk or PSTN. If PSTN than pick Dahdi if SIP than pick SIP.

Can you setup a number that we can test sending a fax to. You can PM a numebr to me if you want. I am curious to know if inbound faxing with spandsp works

__________________

Tony Lewis
Schmooze Com, Inc.
FreePBX Developer

Sorry but I must have

On April 16th, 2011 dcitelecom (tadpole) said:

Sorry but I must have misunderstood the issue. We have a Ast 1.6 system running with Spandsp and from my experience when that is installed properly you can select Nvfax as faxt detection option. In your distro nvfax is greyed out so in my opinion spandsp is not installed correctly.

I am not commenting at all on whether inbound fax works with Dahdi or SIP detection since that should have worked even before your changes. I'll try it next week but please comment on the nvfax option.

nvfax detect is not

On April 16th, 2011 p_lindheimer (Developer) said:

nvfax detect is not span_dsp, it is an independent application whose sole purpose is to detect if something is a fax signal or not and route the call, on SIP channels which traditionally were not supported in Asterisk.

On 1.8, this is available native in Asterisk, so it's not clear whether there is any need for NVFaxDetect though feedback from the users on the reliability of the native SIP fax detect is obviously something we want to hear about.

__________________

Philippe Lindheimer - FreePBX Project Leader
FreePBX Training Opportunities - Click Here
Get Official Paid Support - Click Here

With asterisk 1.8 the SIP

On April 16th, 2011 tonyclewis (Developer) said:

With asterisk 1.8 the SIP Fax Detect is the replacement to NVFax. Prior to 1.8 you had to use a 3rd party application to use fax detect with SIP channels. With later 1.6 and all 1.8 its built into the res_fax which is built into Asterisk 1.8.

You should be good to go just using the SIP settings.

Philippe can we have nvfax disabled with a advanced setting module option so it does not show up even greyed out.

__________________

Tony Lewis
Schmooze Com, Inc.
FreePBX Developer

Google Voice

On April 21st, 2011 giboshi (tadpole) said:

Will Google Voice be part of the GI setup?

Hope to start testing this weekend.

Thanks

No Google voice please. GV

On April 21st, 2011 dcitelecom (tadpole) said:

No Google voice please. GV is the reason why we moved away from PIAF. Their Asterisk 1.8 install with Google voice contains so many changes that it does not even install on our system. It causes an Asterisk segmentation crash on our server. If you want GV there is already a very good distro that caters to your needs. Let's keep the FreePBX distro lean and mean.

For the Record...

On April 28th, 2011 wardmundy (Contributor) said:

We understood the segfault was traced to hostname changes made in the midst of installing Asterisk 1.8 aka PIAF-Purple. There is no Google Voice in PBX in a Flash of any flavor unless you opt to also install Incredible PBX.

SIP Trunk service

On April 21st, 2011 giboshi (tadpole) said:

So if no G-voice, then what is the best and cheapest SIP Trunk service to use. I tried to use Skype but can't get a local number - GV gave me a local number.

There is no such thing as

On April 21st, 2011 tonyclewis (Developer) said:

There is no such thing as good and cheap. If you want a quality trunk I would use the SIPStation FreePBX trunking. There is a module in the distro that will auto setup your trunks for you once you purchase them.

__________________

Tony Lewis
Schmooze Com, Inc.
FreePBX Developer

I am not pointing any

On April 28th, 2011 dcitelecom (tadpole) said:

I am not pointing any fingers and I am not blaming anyone. Fact is purple does not install on some systems (for whatever reason) and it's ok to have certain hardware requirements.

I am simply pointing out that GV already exists on PIAF which is a GOOD distro that caters to this need and thus we don't have to re-invent the wheel and add it to the FreePBX distro as well.

@giboshi. Looks to me like you are looking for a free service not good and cheap. Sipstation is good and cheap. GV is free and ok but I would not use it in a production system where my job depends on the uptime.

I just installed fop2 on the

On May 2nd, 2011 stonet (tadpole) said:

I just installed fop2 on the new distro, when I run panel I get a login window that says extension and password. I only have one extension configured at present so tried this with voicemail and sip passwords. Server then tries to load ten times to no avail. Pretty sure this is not a firewall issue as fop1 was running OK.

Any ideas?

use Voice mail user &

On May 2nd, 2011 ou812 (tadpole) said:

use Voice mail user & password same as when you dial *98

gary

Tried that

On May 2nd, 2011 stonet (tadpole) said:

I tried that but does not connect to the server, however, I have got the panel to run directly using url http://pbx ip address/fop2/?exten=extn number&pass=vm password. It seems to me that something is not happening when I try to run it from the button in frepbx gui.

I have the same issue as

On May 2nd, 2011 LoKeyM4A3 (tadpole) said:

I have the same issue as stonet. I am able to login to FOP2 using the url but when using the PANEL tab it says "One moment please... Connecting to server, attempt number : 1" but it never connects.

FOP2 showing old data

On May 10th, 2011 brk (tadpole) said:

I upgraded to FOP2 and it is showing outdated info and doesn't seam to work. I can't dial an outbound call for an extension (does nothing) and it is showing extensions have voicemail but it has already been listened and deleted.

If I click "refresh labels" on the FOP2 section of the admin panel, then the voicemail info in FOP2 is updated to the current status.

Guys with the latest

On May 2nd, 2011 tonyclewis (Developer) said:

Guys with the latest 1.8.1.2-5 Upgrade we have included native asterisk support for gtalk and jabber. Please see http://www.freepbx.org/forum/freepbx-distro/distro-discussion-help/relea... for more information on the upgrade process.

__________________

Tony Lewis
Schmooze Com, Inc.
FreePBX Developer

For the people that cannot

On May 3rd, 2011 asternic (Developer) said:

For the people that cannot connect via the link, try flushing your browser cache.

__________________

Nicolás Gudiño
Buenos Aires - Argentina

Please identify chargeable

On May 4th, 2011 dcitelecom (tadpole) said:

Please identify chargeable modules as such so we know what's offered commercially. Thanks.

The whole Commercial store

On May 4th, 2011 tonyclewis (Developer) said:

The whole Commercial store and eco system will be announced shorty with a whole portal to log in and purchase hardware, software and support. Just bear with us tell after our OTTS training next week. Right now we have lots of Beta testers testing our different modules and making sure all the bugs are worked out of them since they were all built for FreePBX V2.8 and asterisk 1.4.

__________________

Tony Lewis
Schmooze Com, Inc.
FreePBX Developer

dcitelecom - I am surprised

On May 4th, 2011 SkykingOH (Leap Frog) said:

dcitelecom - I am surprised you did not join us at OTTS, it's almost in your backyard this time.

Is fax2email working for

On May 9th, 2011 dcitelecom (tadpole) said:

Is fax2email working for anyone? I just tried it a couple of times and it failed each time. I got a debug if interested. Just give me an upload address.

fax2email works great. Post

On May 10th, 2011 SkykingOH (Leap Frog) said:

fax2email works great.

Post debug and pastebin.ca

Thanks. I just needed to

On May 10th, 2011 dcitelecom (tadpole) said:

Thanks. I just needed to know that it works so I tinkered around until we got it to work. Turns out we forgot to update postfix with the sender info. Maybe the info could be requested during the install or a reminder could be posted after the install as it's easy to forget.

So now I get faxes with Subject line:
New fax from 15141234567 %3C15145551212%3E

Anyone know how to define or get rid off the %3C and %3E (variables?)

My bad. It's html code %3C =

On May 11th, 2011 dcitelecom (tadpole) said:

My bad. It's html code %3C = "<" and %3E = ">"
but my email client does not speak html in the subject line.
Is there any way not to send the html tags?

how about a 64bit build of

On May 17th, 2011 mustardman (Leap Frog) said:

how about a 64bit build of FreePBX distro and addons?

Not at this time. Maybe in

On May 17th, 2011 tonyclewis (Developer) said:

Not at this time. Maybe in the future

__________________

Tony Lewis
Schmooze Com, Inc.
FreePBX Developer

Wanpipe RPM

On July 22nd, 2011 lanmd (tadpole) said:

Where can you get the wanpip rpm?

SmokePing

On August 12th, 2011 255 (tadpole) said:

What would be the chances of getting a RPM install for smokeping?

I find it very usefull to measure loss and latency to my sip providers.

Would others find this usefull???

Thanks

__________________

Alan Scott
Logical Solutions, NZ

wanpipe

On August 19th, 2011 brk (tadpole) said:

should "yum install wanpipe" work? I found that "yum install wanpipe-2.6.18-194.17.1.el5" installed the code.

But to backup, my goal is to use a UT50 for timing source. Do I even need to be messing with this?

did you ever figure out how

On August 30th, 2011 dcitelecom (tadpole) said:

did you ever figure out how to install the UT50? If yes can you post the instructions?

I'd like to see the

On August 31st, 2011 dcitelecom (tadpole) said:

I'd like to see the wanpipe-voicetime drivers for UT50 added since it's quite popular as a timing device.

Can you try the latest RPM

On August 31st, 2011 tonyclewis (Developer) said:

Can you try the latest RPM for sangoma. It will also require the latest dahdi 2.5. Both are in the yum repos. Just yum update dahdi* wanpipe* should work.

I am not sure if they added support into wanpipe yet for that device. If not than we have to wait on Sangoma to get it into the 3.5 wanpipe which is their current stable release and 3.6 is still not stable.

__________________

Tony Lewis
Schmooze Com, Inc.
FreePBX Developer

Sure. Thanks. I'll try it on

On September 1st, 2011 dcitelecom (tadpole) said:

Sure. Thanks. I'll try it on a test machine.
Won't this invalidate my copy of the distro? I.e. will I have a customized distro version that may not get updated properly via the regular distro updates?

yum update dahdi*

On September 1st, 2011 dcitelecom (tadpole) said:

yum update dahdi* wanpipe*
Loaded plugins: fastestmirror, kmod
Loading mirror speeds from cached hostfile
Excluding Packages in global exclude list
Finished
Setting up Update Process
No Packages marked for Update

You are correct. Those

On September 1st, 2011 tonyclewis (Developer) said:

You are correct. Those packages are only in the latest repos which get enabled with the next upgrade. Sorry I forogt we locked down what packages go into each version to keep people from breaking things with yum updates. It should be moving forward safe inside a release to do yum updates as only packages for that release will be in yum including kernels.

The upgrade should be ready today once we finish the Asterisk 1.8.6 RPM's

__________________

Tony Lewis
Schmooze Com, Inc.
FreePBX Developer

Based on the reply I got

On September 1st, 2011 dcitelecom (tadpole) said:

Based on the reply I got from Sangoma tech support it may be difficult to install a UT50 timing device on a FreePBX distro system because I can't download the "Wanpipe Driver Minimum Requirements" on a distro system. However the UT50 is highly popular as a timing device so maybe you could include the required files in your distro or try to install on one of your systems and post back the results?

FROM SANGOMA:
The voicetime driver is very different from our regular driver because it does
not perform the same functions as our wanpipe driver. The UT50 is
specifically designed only to send a steady clock signal into the USB port,
and our voicetime driver is a lightweight driver which plugs into zaptel/dahdi
and provides the timing source.

In order to install the UT50, you will need to follow the instructions for
installing from source, and download the sources for your currently running
dahdi and kernel, as well as the voicetime source.

The easiest way is to check which dahdi source you are currently using by
running the command "dahdi_cfg -vvv" and verifying with the first few lines of
output. Your kernel development package will contain the kernel sources you
need, which is included in the requirements for a source install
http://wiki.sangoma.com/Wanpipe-Requirements

Yes the UT timing device

On September 1st, 2011 tonyclewis (Developer) said:

Yes the UT timing device does not use wanpipe. It uses its own customer drivers. I kept thinking the USB FXO device that they finally added to the 3.5.x firmware that was only available in the 3.6 for awhole.

We can look at building RPM's for it but its going to be a few weeks before we can get to it.

__________________

Tony Lewis
Schmooze Com, Inc.
FreePBX Developer

OK. Thanks. That's good

On September 1st, 2011 dcitelecom (tadpole) said:

OK. Thanks.
That's good news.

If you have distro version

On September 9th, 2011 dcitelecom (tadpole) said:

If you have distro version 1.8.2.0-2 or higher you can use the instructions
posted below to install the driver.

http://wiki.sangoma.com/Wanpipe-Requirements
(btw it's yum -y install *awk not awk*)
http://wiki.sangoma.com/sangoma-wanpipe-voicetime

Only caveat is that you need to reinstall each time Dahdi is updated.

We are hoping to have RPMs

On September 7th, 2011 tonyclewis (Developer) said:

We are hoping to have RPMs shortly for this and are working with Sangoma to get a spec file built first.

__________________

Tony Lewis
Schmooze Com, Inc.
FreePBX Developer

Module 'app_konference.so' was not compiled with the same compil

On October 4th, 2011 coolname (tadpole) said:

update：
i use the asterisk18-1.8.6.0-1_centos5.src.rpm from http://yum.freepbxdistro.org/pbx/1.8.x.x/i386/RPMS/digium/asterisk/1.8/1... and follow the asterisk18.spec to build the source ,but it report the same error

***********************************************************************************
i use distro to install the asterisk and it works well.But when i want to load a thirdpart module app_konference.so（it compiled with asterisk1.8.6 from asterisk.org with default compile options）,it tells me
[2011-10-03 17:51:08] WARNING[9217]: loader.c:768 inspect_module: Module 'app_konference.so' was not compiled with the same compile-time options as this version of Asterisk.
[2011-10-03 17:51:08] WARNING[9217]: loader.c:769 inspect_module: Module 'app_konference.so' will not be initialized as it may cause instability.
[2011-10-03 17:51:08] WARNING[9217]: loader.c:852 load_resource: Module 'app_konference.so' could not be loaded.
So what's compile-time option of asteriks18 from yum?

asterisk18.i386 1.8.6.0-1_centos5 installed
asterisk18-addons.i386 1.8.6.0-1_centos5 installed
asterisk18-addons-bluetooth.i386 1.8.6.0-1_centos5 installed
asterisk18-addons-core.i386 1.8.6.0-1_centos5 installed
asterisk18-addons-mysql.i386 1.8.6.0-1_centos5 installed
asterisk18-addons-ooh323.i386 1.8.6.0-1_centos5 installed
asterisk18-app_flite.i386 0.6-6 installed
asterisk18-core.i386 1.8.6.0-1_centos5 installed
asterisk18-curl.i386 1.8.6.0-1_centos5 installed
asterisk18-dahdi.i386 1.8.6.0-1_centos5 installed
asterisk18-doc.i386 1.8.6.0-1_centos5 installed
asterisk18-voicemail.i386 1.8.6.0-1_centos5 installed