That’'s right, this one…
http://www.couplet.be/temp/status_fix/asterisk-im.jar
- J
Can you check your log files ?
No change… (did it a third time)
This last time, I rebooted before redownloading the twice modified jar (the new one listed about ^^^^). Also, I made sure to reregister all phones (asterisk & wildfire sitting on same box, phones still think they’‘re registered on reboot). Also turned on debuging. Having looked at it, I really dunno why it can’‘t find “device/jidmapping” as the “Dial a Number” and “Call” buttons do work. Just the asterisk-specific status and incoming call notification popups that don’'t.
- J
DEBUG INFO
2006.08.07 17:16:34 Loading plugin admin
2006.08.07 17:16:46 Loading plugin asterisk-im
2006.08.07 17:16:48 Loading plugin broadcast
2006.08.07 17:16:48 Loading plugin contentfilter
2006.08.07 17:16:48 Loading plugin presence
2006.08.07 17:16:50 Loading plugin registration
2006.08.07 17:16:50 Loading plugin search
2006.08.07 17:16:51 Loading plugin sparkmanager
2006.08.07 17:16:52 Loading plugin userimportexport
2006.08.07 17:16:53 Loading plugin userservice
2006.08.07 17:19:33 Connect Socket[addr=/192.168.16.222,port=4392,localport=5222]
2006.08.07 17:19:59 Connect Socket[addr=/192.168.16.223,port=1594,localport=5222]
2006.08.07 17:22:12 Asterisk-IM: Processing NewState:UP event channel : SIP/207-779f id: 1154985726.0
2006.08.07 17:22:12 Asterisk-IM OnPhoneTask: Could not find device/jid mapping for device SIP/207 returning
2006.08.07 17:22:32 Asterisk-IM HangupTask not active SIP/Teliax
2006.08.07 17:22:32 Cannot destroy non-existent CallSession with id: 1154985728.1
2006.08.07 17:22:32 Asterisk-IM HangupTask not active SIP/207
2006.08.07 17:22:36 Asterisk-IM RingTask called for user PhoneUser{id=3, username=’‘User1’’}
2006.08.07 17:22:41 Asterisk-IM: Processing NewState:UP event channel : SIP/240-e803 id: 1154985756.3
2006.08.07 17:22:41 Asterisk-IM OnPhoneTask: Could not find device/jid mapping for device SIP/240 returning
2006.08.07 17:22:41 Asterisk-IM: Processing NewState:UP event channel : SIP/207-5016 id: 1154985754.2
2006.08.07 17:22:41 Asterisk-IM OnPhoneTask: Could not find device/jid mapping for device SIP/207 returning
2006.08.07 17:22:51 Asterisk-IM HangupTask not active SIP/240
2006.08.07 17:22:51 Asterisk-IM HangupTask not active SIP/207
2006.08.07 17:23:03 Asterisk-IM: Processing NewState:UP event channel : SIP/240-a50f id: 1154985781.4
2006.08.07 17:23:03 Asterisk-IM OnPhoneTask: Could not find device/jid mapping for device SIP/240 returning
2006.08.07 17:23:03 Asterisk-IM RingTask called for user PhoneUser{id=4, username=’‘User2’’}
2006.08.07 17:23:05 Asterisk-IM: Processing NewState:UP event channel : SIP/207-cdf8 id: 1154985783.5
2006.08.07 17:23:05 Asterisk-IM OnPhoneTask: Could not find device/jid mapping for device SIP/207 returning
2006.08.07 17:23:07 Asterisk-IM HangupTask not active SIP/207
2006.08.07 17:23:07 Asterisk-IM HangupTask not active SIP/240
2006.08.07 17:23:22 Asterisk-IM: Processing NewState:UP event channel : SIP/mycompany-e9a8 id: 1154985802.6
2006.08.07 17:23:22 Asterisk-IM OnPhoneTask: Could not find device/jid mapping for device SIP/mycompany returning
2006.08.07 17:23:29 Asterisk-IM RingTask called for user PhoneUser{id=3, username=’‘User1’’}
2006.08.07 17:23:37 Asterisk-IM: Processing NewState:UP event channel : SIP/240-aef5 id: 1154985809.7
2006.08.07 17:23:37 Asterisk-IM OnPhoneTask: Could not find device/jid mapping for device SIP/240 returning
2006.08.07 17:23:52 Asterisk-IM HangupTask not active SIP/240
2006.08.07 17:23:52 Asterisk-IM HangupTask not active SIP/mycompany
2006.08.07 17:23:52 Cannot destroy non-existent CallSession with id: 1154985802.6
INFO
2006.08.07 17:16:47 Initializing phone plugin
2006.08.07 17:16:47 Initializing Asterisk Manager connection
2006.08.07 17:16:47 Connecting to localhost port 5038
2006.08.07 17:16:47 Connected via Asterisk Call Manager/1.0
2006.08.07 17:16:47 Successfully logged in
2006.08.07 17:16:47 Determined Asterisk version: Asterisk 1.2
2006.08.07 17:16:47 Adding new queue 202
2006.08.07 17:16:47 Adding new queue 201
2006.08.07 17:16:47 Adding new queue 200
2006.08.07 17:16:47 Adding new queue default
2006.08.07 17:16:47 Registering phone plugin as a component
2006.08.07 17:22:06 Adding channel SIP/207-779f
2006.08.07 17:22:08 Adding channel SIP/Teliax-815f
2006.08.07 17:22:12 Adding channel SIP/Teliax-815f
2006.08.07 17:22:12 OnPhoneTask called for user PhoneUser{id=4, username=’‘User2’’}
2006.08.07 17:22:12 no sessions
2006.08.07 17:22:12 Linking channels SIP/207-779f and SIP/Teliax-815f
2006.08.07 17:22:32 Unlinking channels SIP/207-779f and SIP/Teliax-815f
2006.08.07 17:22:32 OnPhoneTask: Could not find device/jid mapping for device SIP/Teliax returning
2006.08.07 17:22:32 Removing channel SIP/Teliax-815f due to hangup
2006.08.07 17:22:32 OnPhoneTask called for user PhoneUser{id=4, username=’‘User2’’}
2006.08.07 17:22:32 no sessions
2006.08.07 17:22:32 Removing channel SIP/207-779f due to hangup
2006.08.07 17:22:34 Adding channel SIP/207-5016
2006.08.07 17:22:36 Adding channel SIP/240-e803
2006.08.07 17:22:36 Adding channel SIP/240-e803
2006.08.07 17:22:41 OnPhoneTask called for user PhoneUser{id=3, username=’‘User1’’}
2006.08.07 17:22:41 no sessions
2006.08.07 17:22:41 OnPhoneTask called for user PhoneUser{id=4, username=’‘User2’’}
2006.08.07 17:22:41 no sessions
2006.08.07 17:22:41 Linking channels SIP/207-5016 and SIP/240-e803
2006.08.07 17:22:51 Unlinking channels SIP/207-5016 and SIP/240-e803
2006.08.07 17:22:51 OnPhoneTask called for user PhoneUser{id=3, username=’‘User1’’}
2006.08.07 17:22:51 no sessions
2006.08.07 17:22:51 Removing channel SIP/240-e803 due to hangup
2006.08.07 17:22:51 OnPhoneTask called for user PhoneUser{id=4, username=’‘User2’’}
2006.08.07 17:22:51 no sessions
2006.08.07 17:22:51 Removing channel SIP/207-5016 due to hangup
2006.08.07 17:23:01 Adding channel SIP/240-a50f
2006.08.07 17:23:01 Adding channel SIP/240-a50f
2006.08.07 17:23:03 OnPhoneTask called for user PhoneUser{id=3, username=’‘User1’’}
2006.08.07 17:23:03 no sessions
2006.08.07 17:23:03 Adding channel SIP/207-cdf8
2006.08.07 17:23:03 Adding channel SIP/207-cdf8
2006.08.07 17:23:05 OnPhoneTask called for user PhoneUser{id=4, username=’‘User2’’}
2006.08.07 17:23:05 no sessions
2006.08.07 17:23:05 Linking channels SIP/240-a50f and SIP/207-cdf8
2006.08.07 17:23:07 Unlinking channels SIP/240-a50f and SIP/207-cdf8
2006.08.07 17:23:07 OnPhoneTask called for user PhoneUser{id=4, username=’‘User2’’}
2006.08.07 17:23:07 no sessions
2006.08.07 17:23:07 Removing channel SIP/207-cdf8 due to hangup
2006.08.07 17:23:07 OnPhoneTask called for user PhoneUser{id=3, username=’‘User1’’}
2006.08.07 17:23:07 no sessions
2006.08.07 17:23:07 Removing channel SIP/240-a50f due to hangup
2006.08.07 17:23:22 Adding channel SIP/mycompany-e9a8
2006.08.07 17:23:22 OnPhoneTask: Could not find device/jid mapping for device SIP/mycompany returning
2006.08.07 17:23:29 Adding channel SIP/240-aef5
2006.08.07 17:23:29 Adding channel SIP/240-aef5
2006.08.07 17:23:37 OnPhoneTask called for user PhoneUser{id=3, username=’‘User1’’}
2006.08.07 17:23:37 no sessions
2006.08.07 17:23:37 Linking channels SIP/mycompany-e9a8 and SIP/240-aef5
2006.08.07 17:23:52 Unlinking channels SIP/mycompany-e9a8 and SIP/240-aef5
2006.08.07 17:23:52 OnPhoneTask called for user PhoneUser{id=3, username=’‘User1’’}
2006.08.07 17:23:52 no sessions
2006.08.07 17:23:52 Removing channel SIP/240-aef5 due to hangup
2006.08.07 17:23:52 OnPhoneTask: Could not find device/jid mapping for device SIP/mycompany returning
2006.08.07 17:23:52 Removing channel SIP/mycompany-e9a8 due to hangup
Message was edited by: TheShniz
-
After you have rebooted asterisk, you have to reboot your phones too or to manually force you phone to register again. If you don’'t phone will think they are registred (registration before reboot not yet expired) but asterisk after reboot will not have the phone registred until the register expire (usually 3600sec).
-
“Asterisk-IM OnPhoneTask: Could not find device/jid mapping for device SIP/207 returning” mean that there is no user logged on wildfire that has a device of name SIP/207. Please go to your asterisk-im phone mapping screen on wildfire’'s web admin and verify that you have a user logged in that has an device SIP/207 (second column).
Herve
UGHHHHHHHHHHHHHHHhhhhhhhhhhhhhh!!!
::sigh::
yes, it worx now…
You know, the ‘‘lil things’’ can really be a pain… things like non-existant documentation.
Wildfire and Asterisk-IM have many lil nuances that take a lil bit of trial & error before you figure why things don’‘t work, then it’'s just ‘‘d0h, that’‘s kinda retarded… why’‘d they do that?!?’’
Well I just found one more ‘‘retarded thing’’ to add to my list.
Rule # 23597:
You ‘‘should’’ NOT have anything but lowercase usernames.
Reason:
The ‘‘Client Sessions’’ lowercases everything!
i.e.)
User1 is NOT online, but user1 is… solution: rename username & phone mapping!
I have no idea why it works just fine like this in 3.0.0, but it does!
Just 1x of those things, GRRRRrrrrrrrrrrrrrrr…
-
J
SOOOOoooooooooooooooooooooooooooooo…
…let’‘s recap some of the wtf’'s:
(I’'ve already posted in other threads, but again for redundancy/search sake)
Wildfire > Server > Edit Properties > Server Name MUST be identical to Spark > Server.
Either IP or DNS, but they must match precisely. Host name in the Asterisk does NOT have to match what’'s in the DNS server, but Spark must match Wildfire.
Wildfire > Asterisk-IM
a.) Does NOT like anything EXCEPT from-sip in Asterisk Context
(i.e. can’'t specify from-internal)
b.) Can NOT leave Asterisk Context or Default Caller ID blank, MUST specify
c.) Can NOT use spaces in Default Caller ID
from-sip is NOT a valid context in Trixbox
Add to /etc/asterisk/extensions_custom.conf:
include => from-internal
You ‘‘should’’ NOT have anything but lowercase usernames.
Reason: The ‘‘Client Sessions’’ lowercases everything!
No matter if the user is logged in and working or not, if it’'s not lowercase, the status will NOT update!
i.e.)
User1 is NOT online, but user1 is… solution: rename username & phone mapping!
SOoooooooooooooooooo…
How ''bout ''dem ''der incoming call notifications, any idea?
WHOOOOAAAaaaaaaaaaaaaaaaaaaaaaaa
Was gathering notes & information on the subject, then I noticed…
Inbound Call Popup Notifications ARE WORKING NOW ALSO!!!
P.S.
Read my above wtf’'s recap and see if you can get your notifications working also, lemme know if you need help!
Oh my, whatever will I do w/ a 100% working Wildfire setup…
…HMmmmmmmm, time to stress test?
- J (extremely happy right now!)
Concerning your remark :
“a.) Does NOT like anything EXCEPT from-sip in Asterisk Context (i.e. can’'t specify from-internal)”
I think this depends on your asterisk configuration. Currently the context I use is “from-ip-phones” that is a custom one I configured in Asterisk. That configuration is working fine here.
You’‘re probably right on that as ‘‘from-sip’’ would be considered as a custom context in my (see also: Trixbox’‘s) configuration. I’'ve tried other ‘‘standard/default’’ contexts w/ none working, although I never tried a custom one (other than the expected ‘‘from-sip’’). So that makes sense…
…so have you got your notifications working?
- J
yep, I have notification working. May be I will add a url popup to display a sugarCRM contact.
Very good… I’‘d offer help to adding a URL popup such as SugarCRM (as I’‘ve had several requests for this already!), but you seem to be a lil ahead of me on Wildfire. Anyways, a much requested feature, am here if you’'d like some help!
- J
Thank you for fixing this! Presence and notifications both now working in Spark. Presence working in Exodus. Does Exodus support notifications?
Thanks
Tom