PIAF 1.2 & Asterisk 1.6

[sota]

I'm fairly new to Asterisk, but I wanted to try 1.6 to see if I could get it working with MS Exchange 2007 UM. On a clean install, I found that there was no SIP functionality - I couldn't even run SIP commands from the cli. Looking through the log file, there were lots of error messages about missing #include files - for example most of the files listed in sip.conf did not exist.
Once I created these files, sip functionality was restored and I could register phones OK - I'm still trying to get it talking to Exchange though.

 

[wardmundy]

If you're not seeing SIP commands in the CLI, then it sounds like you don't have an Internet connection to your server. 1.6 runs fine on several machines that I test on including as a VMware image.

 

[sota]

This is the type of error I was getting:
May 2 17:30:37] VERBOSE[2835] logger.c: == Parsing '/etc/asterisk/sip.conf': [May 2 17:30:37] VERBOSE[2835] logger.c: == Found
[May 2 17:30:37] VERBOSE[2835] logger.c: == Parsing '/etc/asterisk/sip_general_additional.conf': [May 2 17:30:37] VERBOSE[2835] logger.c: == Found
[May 2 17:30:37] ERROR[2835] config.c: The file 'sip_general_custom.conf' was listed as a #include but it does not exist.
[May 2 17:30:37] NOTICE[2835] chan_sip.c: Unable to load config sip.conf

The PIAF box connects fine to the internet and I can ping internal & external hosts OK. Like I said, once I created the missing .conf files and reloaded asterisk, the sip commands became available from the cli. I tried a couple of clean installs with the same result - the only other possibility I can think of is a bad ISO or CD.

 

[darmock]

of course you went into freepbx and tapped the orange reload bar correct? fpbx creates a number of files after you tap the orange reload bar the first time you log in. this might be where you problem is

tom

 

[sota]

Well, I downloaded a fresh ISO and confirmed the MD5 checksum and burned a new CD. I installed using ks16alt and ran update-scripts16, update-fixes16 and genzaptelconf. Went back into FreePBX and clicked on reload.

But when I goto the Asterisk cli, there are still no sip commands available. If I look in /var/log/asterisk/full - there are still lots of error messages about missing #include conf files.

 

[wardmundy]

Have you read the Installation Guide? You have to install the FreePBX modules separately. That populates some of the config files for Asterisk.

 

[sota]

I had gone through the Installation Guide, but it makes no mention of updating FreePBX. I just found the info on another site - nerdvittles.com - and that seems to have solved the problem.

Thanks

 

[darmock]

well depending on when you ran update-fixes16 it might have been the patch that I pushed out earlier today to deal with this.
Update-fixes16 add fix #103 and I recommend you run update-scripts16 and update-fixes16 again to get the very latest stuff. Please note that if you have created the missing conf files and added anything to them update-fixes16 will overwrite them with as many of the proper conf files as I could find.

I applied this patch to a running Asterisk 1.6 install (with sip,zaptel, and iax trunks) and had no problems.... your mileage will vary. I have also patched the load file for asterisk 1.6 which should be available later this evening.

This patch should get rid of the errors in the full log file although my 1.6 systems seemed to run just fine without any of these files.


Tom

 

[jroper]

Hi

In 1.4 there are all sorts of dire warnings of things that will go wrong if there is a #include statement, and the included file does not exist.

Asterisk 1.6 must have implemented this rule. When we install FreePBX in the install script, we only create the files that are not created by FreePBX to avoid any conflicts.

Thus, when the install is completed, The first thing to do is log into FreePBX for the first time and click the orange reload bar, which goes and creates all the files required.

Hopefully, the FreePBX team has set all the files to be created where they have put in a #include.

Check /var/log/asterisk/full for this kind of error, and if there are any files not created, then let us or the FreePBX team know.

Joe

 

[sota]

As I'm only testing 1.6 at the moment, it's no problem to do a reinstall. I've noticed that the first time I go into FreePBX, I don't see the orange reload bar, but as I mentioned above, downloading and installing the additional modules seems to solve the problem.
Initially, I had not realised that Asterisk required #include files to actually exist, it was only by careful reading of /var/log/asterisk/full that I realised what was going on.

Pat Rooney

 

[darmock]

I also included the patch that presses the orange reload bar for the first time on new systems thus the requirement to press the reload bar the very first time has gone away.


I do my utmost to engineer out the human factor in these complex system as this is usually the factor that causes things to break.

Tom

 

[foneman]

Sweet Install

Guys,
Thanks so much. I just did the first install onto a lab box of 1.2 and went SMOOOOOTHHHHHH....especially on one of those cheap VIA boards....
Not to but any progress on FAX???

 

[darmock]

In short No!!!!!!

hehe I wish I had more time but this is really at the bottom of the list to run on PIAF I am afraid. Some of my testing of late will cause a walmart special box to slow down to nothing while a fax is inbound.

My thoughts are that this needs to go on a separate server and people need to bite the bullet and by an external modem. Then simply output the inbound fax call thru a digium card to the modem which answers and spools off of another server. This is what I do at my office.

While Spandsp and app_fax seems to be working under ast 1.6 is still causes the occasional core dump which means I won't put it on any of my servers! It is somewhat more problematic under Ast 1.4.......

Now some have reported it works flawlessly... why can't you make it work... you must be too stupid to own a computer.... etc. My thoughts are this is a fairly complex process which is prone to breakdowns and if it breaks down it takes your phone system with it.

From my clients perspective this is unacceptable. So my secret is out! I generally only develop stuff for PIAF that I use and my clients want.... gee could it be they are paying me to do this?

Since none of them seem to want to fax stuff it gets put to the very very bottom of the development list. I revisit it every so often to see if things have become more stable running under asterisk.

I have put up a few servers using hylafax/avantfax on separate servers and they run flawlessly.


Tom

 

[foneman]

I understand completely!

I am sorry to have hit a sore spot... I didnt mean to.
I definately understand needing to do what the customer's want as they pay the rent. I am in the same boat.
I was just asking. I too am trying on my own to play with it like Ed did just like you, no time...need 30 hrs in a day.
Again, I cannot thank you enough for what you have done.
Gary

 

[darmock]

hehe mostly my response was tounge in cheek..... Fax stuff is still a work in progress and I cant see that changing anytime soon. i have looked at the other distros that are supposed to have it working. they all suffer from core dumps and other oddities. I now expect some fanboy to tell me I am wrong etc etc... sigh

tom

 

[cosmicwombat]

HEY I am a fanboy...

oh....wait... wrong forum.....

Actually, I have FAX working on my PiaF 1.1 system fine. I have one POTS to a TDM 400 and detection works. I have three IAX modems and outbound works too.

Robert

 

[darmock]

I just knew you would pick up on that...... hehe couldn't resist. Myself I just don't have time to even try to support it. I have been having a blast getting spandsp, app_fax_rx and tx, and NV_background working under 1.6. What is even more interesting is digium has modified the app_fax driver and made inroads to actually making it work on asterisk 1.6.

Tom