Used Google Voice module and now outgoing says all circuits busy

[wardmundy]

This is a double or triple post. zz: One post is sufficient! It's already been addressed in another one of your threads.

 

[zzvikingzz]

I'm not looking to cause trouble here. I'm trying to get help. The other post was related to re-install and then enabling of the NEWLY supported version of GV module. There is a problem here for me on enabling the NEWLY supported version of GV module on ISO 1.7.5.6.2 install. It changes config for outbound calls after enabling after this ISO install. I was trying to find out how to repair or change what the install does so I didn't have to rebuild. My apologies for any confusion.

I have both ISOs (1.7.5.6.2 & 1.7.5.6.3) installed now on 2 different machines. Will test on Newly supported version on 1.7.5.6.3 and leave it disabled on 1.7.5.6.2. The Marcus Brown module should be removed from the install on both for Purple to avoid confusion.

 

[wardmundy]

The payload including the Google Voice module is identical in 1.7.5.6.2 and 1.7.5.6.3 with the exception that 1.7.5.6.3 now includes an Asterisk 10 option: PIAF-Red. With only one person reporting confusion, it's unlikely we'll be removing the Google Voice module from PBX in a Flash installs. We've documented the appropriate install method in this week's Nerd Vittles article.

 

[zzvikingzz]

followed the article and I'm getting this in the logs after setting the ext up with gv module and inbound route.

[2011-08-29 21:30:39] NOTICE[24346] chan_sip.c: Failed to authenticate on REGISTER to '[email protected]' (Tries 3)
[2011-08-29 21:30:39] NOTICE[24346] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #12)
[2011-08-29 21:30:59] NOTICE[24346] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #49)
[2011-08-29 21:30:59] NOTICE[24346] chan_sip.c: Failed to authenticate on REGISTER to '[email protected]' (Tries 3)

 

[tm1000]

None of those messages are useful.

Edit: Why are you replying in two completely different threads about the same exact topic! UGH!

I give up. White flag raised.

 

[chemcat9]

Module updated to 0.5.0; here is what is in the log file (hoping this is just a configuration error on my part). Should I delete and recreate the google voice entry or will the update pick up and run with the prior entry?

Log:

[2011-08-28 12:12:31] VERBOSE[3205] pbx.c: -- Executing [s@macro-dialout-trunk:27] Dial("SIP/1002-00000000", "local/1112225555@googlevoice-MYGOOGLEid,300,") in new stack
[2011-08-28 12:12:31] VERBOSE[3205] app_dial.c: -- Called local/1112225555@googlevoice-MYGOOGLEid
[2011-08-28 12:12:31] VERBOSE[3206] pbx.c: -- Executing [1112225555@googlevoice-MYGOOGLEid:1] Dial("Local/1112225555@googlevoice-MYGOOGLEid-cde1;2", "Gtalk/MYGOOGLEid/[email protected],tr") in new stack
[2011-08-28 12:12:31] WARNING[3206] channel.c: No channel type registered for 'Gtalk'
[2011-08-28 12:12:31] WARNING[3206] app_dial.c: Unable to create channel of type 'Gtalk' (cause 66 - Channel not implemented)
[2011-08-28 12:12:31] WARNING[3206] app_dial.c: Invalid timeout specified: 'tr'. Setting timeout to infinite
[2011-08-28 12:12:31] VERBOSE[3206] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)
[2011-08-28 12:12:31] VERBOSE[3206] pbx.c: -- Executing [1112225555@googlevoice-MYGOOGLEid:2] NoOp("Local/1112225555@googlevoice-MYGOOGLEid-cde1;2", "GoogleVoice Call to 1112225555 failed") in new stack
[2011-08-28 12:12:31] VERBOSE[3206] pbx.c: -- Auto fallthrough, channel 'Local/1112225555@googlevoice-MYGOOGLEid-cde1;2' status is 'CHANUNAVAIL'

Thanks again (especially for the quick replies)!





tm1000
Guru

Join Date: Dec 2009
Location: Rancho Cucamonga, CA
Posts: 890
Please look in /etc/asterisk/modules.conf and make sure the top part looks like this, you do this after you setup the google voice module:

Code:
[modules]
autoload=yes
load = res_jabber.so
load = chan_gtalk.so
;You must change noload to load on the above 2 lines above AFTER you have configured the conf files!
__________________
Developer: FreePBX/2600hz/Provisioner/Endpoint Manager/Google Voice/Swiss Army Knife/CID Superfecta

Modules.conf was configured as above, I did an amportal restart. Calls still do not route through google voice; logs follow:

[2011-08-30 13:23:51] WARNING[30829] channel.c: No channel type registered for 'Gtalk'
[2011-08-30 13:23:51] WARNING[30829] app_dial.c: Unable to create channel of type 'Gtalk' (cause 66 - Channel not implemented)
[2011-08-30 13:23:51] WARNING[30829] app_dial.c: Invalid timeout specified: 'tr'. Setting timeout to infinite​

Should I have re-entered my google voice information after the mod update?

Thanks!

Purple 1.8 install:

PIAF ver. 1.7.5.6
FreePBX ver. 2.8.1.4
Asterisk ver. 1.8.5.0
Libpri ver. 1.4.12​

 

[MichiganTelephone]

To those who are having issues, I'm just going to throw out a couple things here.

First, please note that an orange bar reload in FreePBX does NOT always catch updated Google Voice configuration information. So before you panic and assume it's not working, try restarting Asterisk. Go into the Asterisk CLI and do "core restart when convenient" and when the system is not handling any calls it will restart Asterisk, and that may get things working. Failing that, there's always a full system reboot, which of course will restart EVERYTHING. I know you are not supposed to ever have to do that in Linux, but I have found by experience that sometimes weird problems clear right up after a reboot, particularly where Asterisk is involved.

Second, the main configuration files that are created or messed with by the various methods of configuring Google Voice are these (in the /etc/asterisk directory):

gtalk.conf
jabber.conf

One thing I would certainly try if things aren't working is moving those files out of /etc/asterisk to another location temporarily, then uninstalling and reinstalling the Google Voice module (please read this entire post before reinstalling). After reinstalling, check to see if those two files have been recreated. The reason for doing that is if you have used a previous method for getting Google Voice to work, there could be conflicting data in those files, and you don't want that. Also, between uninstalling and reinstalling the module, I'd go through extensions_custom.conf and look for any additions you may have made to that file while using an older method, and delete anything you find in there relating to Google Voice (you may also wish to backup that file first, just in case). And also, if you created any custom trunks for use with an older method, you'll want to delete those as well.

The idea basically is to uninstall the Google Voice module, purge everything you can find on the system related to Google Voice (but back up any files before deleting them, in case you REALLY mess something up), and then reinstall the latest version of the Google Voice module. Hopefully at that point it will create a "clean" configuration that works. If that doesn't work you can always return the backup files to their original locations (although if Google Voice isn't working at all, I'm not sure what difference it would make).

Finally, if it's a new Google Voice account, don't forget that it may not work in Asterisk until you've placed at least one outgoing call from the associated Gmail account page (using the "call" icon in the left hand column), and also don't forget to make the Google Voice destination Google Chat (unless you are using a DID to bring in the calls).

And if all else fails, well, there's always starting from scratch with the latest PiaF ISO file , but let's not even think thoughts like that until all other avenues have been exhausted.

 

[fizadmin]

Been struggling with Google Voice - my issue is that Google Voice simply doesn't place any calls. I get the same "All Circuits Are Busy Now" error.

I've updated the GV module to 0.5.0, I've made sure those two modules load and did am portal restart.

This is the error I now see in the logs:

[2011-12-22 10:10:27] NOTICE[11275] chan_local.c: No such extension/context [email protected]@gmail.com while calling Local channel

I'm going try the MT suggestion of completely uninstalling and reinstalling the GV module, and see how that works.

 

[fizadmin]

That did it - set up the appropriate dial plans, and now my US, Canada, and toll-free calls all go out over GV.

Sweet!

Just no caller ID rewriting with outgoing GV calls, but I understand that is no longer possible because of the usual abuse by idiots (probably telemarketers). I can live with that. Probably do some more dial-planning so that "my" calls go out over Vitelity, properly Caller ID'd, and less relevant calls (toll-free, etc..) goes out over GV.