SUGGESTIONS Context=from-internal

[theo69]

Hi,

I am trying to connect two boxes - and have done so succesfully in the past a few times. This time however I am not winning. Box A calls out to Box B. Box B is meant to pass that to the right trunk for outgoing calls. However, even though I have specified context=from-internal, it comes through are from-trunk and therefore doesn't pass it on but rather looks for an internal extension by that number.

Here's what I have in the peer details of my aix2 trunk:

context=from-internal
host=xx.xx.xx.xx
type=user

If I say from-internal, why does it still arrive as from-trunk??????

Thanks a lot for any help

 

[theo69]

Some follow up - I wanted to have a look at the config file to see what it actually has there. Where on earth is that info stored? There is nothing in Iax_additional_custom.conf, nor in iax_additional.conf. This is PIAF 1.3 32 bit.

 

[jmullinix]

Depending on which version of Asterisk you are running, you may need to add

requirecalltoken=no

in your trunk setup.

If your context is from-internal and your calls are arriving as from-trunk, I would say that the call is not being handled by the IAX trunk and is being handled by guest user.

Watch your CLI during a call. That should shed some light.

 

[theo69]

Hi John,

Thanks for the reply. It does indeed seem that the call is not handled by the iax trunk - which again points at the fact that the additional-conf file is simply empty. It appears that it is not written to that file. What could be the cause of that and how would I fix that?

Thanks

Theo

 

[blanchae]

I would check the naming that you've used for your trunk. The name may already be defined and the dialplan isn't getting to your trunk. Names should be unique like IAXtrunkCalif...

 

[theo69]

Hi,

The name is fine - but again, there is no record of the trunk - or any other IAX trunk for that matter in the config files. Unless I am not looking in the correct files, but I am assuming that they are the same as for SIP (but then IAX of course). The SIP info does all appear in the .conf files.

 

[blanchae]

I guess it would help if you told us the version of PiaF, asterisk and stuff plus your complete trunk config. It appears to be just half a config: the user only, what about the peer portion? secrets, etc...?

Follow these instructions on creating an IAX2 trunk between two PBXs. Make sure you use the spreadsheet and fill in the green areas. The configs for both PBXs will be instantly created. I'll have to add "requirecalltoken=no" in as soon as I post this reply

In your case, you would change from-trunk to from-internal on box B or both, your choice.

 

[theo69]

HI Blanchae,

I have followed those instructions literally tens of times before I eventually came here for help, among other ones. I did mention in my first post that I use Piaf 1.3 (Asterisk 1.4). What nobody so far seems to suspect as an issue is the fact that whatever I set up, nothing appears in IAX_additional.conf, nor in any other IAX config file I can find. Am I wrong to see that as the possible cause of it not working? Should it not have written the details into that file (as per the link you gave me?). As I said, I have had several setups with boxes interconnected and have never had an issue before so I am pretty sure that my details are correct. It is only in this particular case that I cannot seem to get it to work.

 

[dswartz]

Not writing to that file seems suspect yes. I assume the permissions on that file are correct? Maybe a reinstall is needed

 

[Alex728]

as a last resort before you reinstall try update-scripts and update-fixes. I've just had a look in iax_additional.conf on my working systems and there is indeed a bunch of stuff in there.

A lot of these fixes appear to change the permissions errors that can plague integrating Asterisk and FreePBX..

 

[theo69]

Hi,

Thanks. A Reinstall is not really an option right now - this is a production server serving 200,000 minutes a month. It is due to be replaced in a few months but not as yet. For the same reason I am very hesitant to run any update-source and update-fixes command. I cannot afford for the server to go down right now. Best bet I suppose is to check permissions, but not entirely sure how to do that and what to check for. Any hints?

Thanks a lot

 

[blanchae]

I would recommend the update-fixes script but not the update-source. update-source can break things, update-fixes fixes bugs.

For checking permissions, run the "ls -l" command on the directory. If you are not familar with that command then you should read Jim Roper's Conversational Linux for Windows users. (pdf)

 

[Bitnetix]

One thing I'd do, but this is me and I grew up in the 20th century:

(not going to bother to make this pretty or complete):

# mysql asterisk
mysql> select * from iax;

Make sure you have _something_ in there. If you do, then you have data in the database, but it is not being pulled during the FreePBX reload to populate the database files.

I'm thinking protections, or perhaps you don't have the IAX module loaded for some reason:

# asterisk -rvvv
*CLI> module show like iax
Module                         Description                              Use Count
chan_iax2.so                   Inter Asterisk eXchange (Ver 2)          0
1 modules loaded

If this comes back with anything other than something similar to what I've shown, then Asterisk doesn't think it needs to load IAX for some reason. The next phase is to figure out why.

 

[theo69]

Hi Guys,

The update fixes script seems to have sorted it out - The iax_additional_conf file now has the detail I was looking for and it works!

Thanks for taking the time to answer the queries.

 

[johnny2000]

I have the same exact problem, however the update-fixes did not solve the problem. I am using piaf 2.0.6, freepbx 2.11, asterisk 11.10. The file iax_additional.conf has all the iax-trunks listed and the context is from-internal. However during the calls, in the asterisk CLI, the call is from-trunk instead of from-internal.

 

[wardmundy]

@johnny2000 Take a look at this thread: http://pbxinaflash.com/community/in...ble-pbx-12-and-asterisk-13.17973/#post-115087

 

[johnny2000]

i will try it