Changed to Asterisk 12, and get xmpp errors

I figured while I was changing things around, and updating my home PBX to FPBX 12, I might as well try jumping the asterisk version to 12 as well, and see how that went. It all seems to be running, but looking at the asterisk full log, I see a constant stream of a res_xmpp error, that for sure will end up making the log huge on the system as it keeps on writing to the log.

I am getting the following error in the log:

[2014-11-11 23:53:14] WARNING[4250] res_xmpp.c: Parsing failure: Hook returned an error.
[2014-11-11 23:53:14] WARNING[4250] res_xmpp.c: Parsing failure: Hook returned an error.
[2014-11-11 23:53:14] WARNING[4250] res_xmpp.c: Parsing failure: Invalid XML.
[2014-11-11 23:53:14] WARNING[4250] res_xmpp.c: Parsing failure: Invalid XML.
[2014-11-11 23:53:14] WARNING[4250] res_xmpp.c: JABBER: socket read error
[2014-11-11 23:53:14] WARNING[4250] res_xmpp.c: JABBER: socket read error

This didn’t happen with Asterisk 11, so not sure what changed, but using the switch command to jump to version 12 left the system with this issue. I will try another version jump, to see what happens, but if anyone knows a fix for this I would love to know. I have tried enabling and disabling the XMPP module in FPBX, it didn’t seem to change anything…

As a follow up, I tried jumping to asterisk 13, same issue. So I dropped back to asterisk 11, and now it has the same issue. Something changed when running the asterisk-version-switch command, that has ticked off xmpp, and we don’t even use/license it.

Any ideas on how to fix this, would sure be helpful…

This actually has nothing to do with the commercial FreePBX module called xmpp. The errors above are from res_xmpp which is not related to the commercial module. Instead it’s releated to the google voice / motif module.

Thanks Andrew, guess I can remove that one, as I think GV doesn’t really even work with the PBX anymore, or I seem to recall seeing news about them disallowing non Google clients. Strange this cropped up as part of the changes, so something must have tickled the issue with the changes…

As a follow up, I went into FPBX and disabled the GV/Motif module, and then made sure no trunks/routes were present that would use it. Still even after a restart of the PBX, I get a constant scroll of the error in the asterisk full log, so I then just performed an “module unload” and that stopped it.

Of course on a restart of the system it will come back, so any ideas on how to fix this issue, or why it’s doing it? Just strange doing the asterisk-version-switch kicked this issue into high gear, but as mentioned even dropping back to 11 didn’t make it go away…

Do you have an account setup through GV/Motif? Removing the trunks and or routes is completely meaningless.

You are really over complicating this whole affair. Go to admin > Asterisk Modules. Add “” to “Excluded Modules”

Thanks Andrew, that must be a new module (asterisk modules) in FPBX, as I don’t recall ever seeing the options to exclude modules via FPBX. So thanks for pointing that new feature out to me, it’s appreciated and I have added the xmpp module to the excluded list…

If you are using motif/GV Trunk and getting this error.

[2015-03-20 20:52:49] WARNING[22432]: res_xmpp.c:3570 xmpp_client_receive: Parsing failure: Invalid XML.
[2015-03-20 20:52:49] WARNING[22432]: res_xmpp.c:3637 xmpp_client_thread: JABBER: socket read error
[2015-03-20 20:52:50] WARNING[22432]: res_xmpp.c:3573 xmpp_client_receive: Parsing failure: Hook 

I instantly corrected this by following these instructions

Re: Problems with Google Voice [SOLVED]
So, it turns out that there's a security setting
 that you have to enable on your Google account for it to work, at least for me. I did this on After doing so, I was able to connect without a problem, and made my first call from my laptop through to my cell phone! 

As the commentator asked why this setting would be necessary/

It is necessary as the default res_xmpp uses plaintext authentication. There are now patches that can change the authentication to a more secure method but they are not available as rpms.

and if anyone asks they won’t be available as RPMS because no one has submitted the work to Digium and we don’t like making patches to Asterisk, especially patches that are unstable as one can see from the DSL reports threads.