I have had a working asterisk 1.4 setup for few years, I upgraded to asterisk 1.8 as per the posts I linked above to get GV working.
I have successfully migrated from 1.4 to 1.8 all of my sip channels and trunks are working properly. I can call between my local sip channels as well as call in and out of my trunks such as freephoneline.ca, etc are working no problem.
After following the wiki i pointed to in my post, I get the following error when I call out using my googlevoice setup on my asterisk.
== Using SIP RTP CoS mark 5
-- Executing [xxxxxxxxxx@internal:1] Dial("SIP/100-00000000", "Motiffirstname.lastname@example.org,,r") in new stack
[Jul 1 19:05:16] WARNING: channel.c:5711 ast_request: No channel type registered for 'Motif'
[Jul 1 19:05:16] WARNING: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'Motif' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
I seems that I don't have the chan_motif.so file loaded. When I try to load it I get the following message.
Hello*CLI> module load chan_motif.so
Unable to load module chan_motif.so
Command 'module load chan_motif.so' failed.
[Jul 1 19:07:16] WARNING: loader.c:418 load_dynamic_module: Error loading module 'chan_motif.so': File not found
[Jul 1 19:07:16] WARNING: loader.c:894 load_resource: Module 'chan_motif.so' could not be loaded.
Where is this file located normally, can I just copy and paste this file.
My Setup: Linksys E3000 Router, Brianslayer Build 14929, Asterisk 1.8
So i have figured out that I need to Asterisk 11 installed in order to use the chan_motif.so module.
My next issue if someone can help with is following as I dial out I get the following error, I have enabled icesupport=yes in the rtp.conf file.
Connected to Asterisk 11.7.0 currently running on Hello (pid = 31162)
== Using SIP RTP CoS mark 5
-- Executing [xxxxxxxxxx@internal:1] Dial("SIP/100-00000002", "Motifemail@example.com,,r") in new stack
-- Called Motiffirstname.lastname@example.org
[Jul 3 05:09:01] ERROR[C-00000002]: chan_motif.c:821 jingle_add_google_candidates_to_transport: Unable to add Google ICE candidates as ICE support not available or no candidates available
-- Motifemail@example.com is proceeding passing it to SIP/100-00000002
So I figured out that I need to have a stun server in the rtp.conf in order to get past my last error of ice support no available.
I next then had an issue with incoming calls, my outgoing calls were working but my incoming calls were not working. So that was fixed by adding the following to modules.conf "noload => res_timing_pthread.so" that fixed my issue with incoming calls all together (my GV or other DID) incoming calls in Asterisk 11 were not working.
I still have two more issues, it seems like now my GV calls come in just fine, but after the first google calls ends it seems my servers hangs and I cannot make or receive another phone call. Though if I reboot the router all is well again, my other DID incoming calls do not exhibit this behaviour.
Again if some one has gone through this would like to hear how they fixed. I have feeling I am either loading too many modules and don't know which ones I really need.
Problem encounter in my last post was resolved with having the right modules loaded, now I don't really know which once are right but I have a working configuration where my local extensions, DID, sip channel and GoogleVoice are working as implemented.