Anyone using Asterisk-IM with asterisk 1.6? It all seems good except for the incoming callerid popups. Worked fine in asterisk 1.4. I do know there are some changes in the manager API for 1.6. Anyone have some ideas? Thanks! --Fred
Oh and heres the manager API output for a typical call if someone wants to look at it and see the diferneces to the older API. Some of the obvious changes spotted ar that CallerID: from 1.4 in most cases becomes CallerIDNum: in 1.6 and Event: Link becomes Event: Bridge. Again thanks for any input anyone might have.
Event: Newchannel
TimeStamp: 2008-10-15 17:30:58
Privilege: call,all
Channel: DAHDI/1-1
ChannelState: 4
ChannelStateDesc: Ring
CallerIDNum: 0000000000
CallerIDName:
AccountCode:
Uniqueid: 1224113458.571
Event: Newexten
TimeStamp: 2008-10-15 17:30:58
Privilege: dialplan,all
Channel: DAHDI/1-1
Context: default
Extension: 0000000000
Priority: 1
Application: Macro
AppData: didexten,SIP/103,103,20
Uniqueid: 1224113458.571
Event: Newexten
TimeStamp: 2008-10-15 17:30:58
Privilege: dialplan,all
Channel: DAHDI/1-1
Context: macro-didexten
Extension: s
Priority: 1
Application: Wait
AppData: 2
Uniqueid: 1224113458.571
Event: NewCallerid
TimeStamp: 2008-10-15 17:30:58
Privilege: call,all
Channel: DAHDI/1-1
CallerIDNum: 0000000000
CallerIDName: XXXXXXXXXXXXXXXXX
Uniqueid: 1224113458.571
CID-CallingPres: 3 (Presentation Allowed, Network Number)
Event: Newstate
TimeStamp: 2008-10-15 17:31:00
Privilege: call,all
Channel: DAHDI/1-1
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 0000000000
CallerIDName: XXXXXXXXXXXXXXXX
Uniqueid: 1224113458.571
Event: Newexten
TimeStamp: 2008-10-15 17:31:01
Privilege: dialplan,all
Channel: DAHDI/1-1
Context: macro-didexten
Extension: s
Priority: 11
Application: Dial
AppData: SIP/103,20,twk
Uniqueid: 1224113458.571
Event: Newchannel
TimeStamp: 2008-10-15 17:31:01
Privilege: call,all
Channel: SIP/103-08339fd0
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Uniqueid: 1224113461.572
Event: ChannelUpdate
TimeStamp: 2008-10-15 17:31:01
Privilege: system,all
Channel: SIP/103-08339fd0
Uniqueid: 1224113461.572
Channeltype: SIP
SIPcallid: 67fc4c486e71c82e4779e21b4c346d2e@172.27.1.150
SIPfullcontact: sip:103@172.27.1.156:5060
Event: ChannelUpdate
TimeStamp: 2008-10-15 17:31:01
Privilege: system,all
Channel: SIP/103-08339fd0
Channeltype: SIP
SIPcallid: 67fc4c486e71c82e4779e21b4c346d2e@172.27.1.150
SIPfullcontact: sip:103@172.27.1.156:5060
Peername: 103
Event: Dial
TimeStamp: 2008-10-15 17:31:01
Privilege: call,all
SubEvent: Begin
Channel: DAHDI/1-1
Destination: SIP/103-08339fd0
CallerIDNum: 0000000000
CallerIDName:XXXXXXXXXXXXXXXX
UniqueID: 1224113458.571
DestUniqueID: 1224113461.572
Dialstring: 103
Event: NewCallerid
TimeStamp: 2008-10-15 17:31:01
Privilege: call,all
Channel: SIP/103-08339fd0
CallerIDNum: 5059384795
CallerIDName:
Uniqueid: 1224113461.572
CID-CallingPres: 3 (Presentation Allowed, Network Number)
Event: Newstate
TimeStamp: 2008-10-15 17:31:01
Privilege: call,all
Channel: SIP/103-08339fd0
ChannelState: 5
ChannelStateDesc: Ringing
CallerIDNum: 0000000000
CallerIDName:
Uniqueid: 1224113461.572
Event: ChannelUpdate
TimeStamp: 2008-10-15 17:31:04
Privilege: system,all
Channel: SIP/103-08339fd0
Channeltype: 1224113461.572
Uniqueid: SIP
SIPcallid: 67fc4c486e71c82e4779e21b4c346d2e@172.27.1.150
SIPfullcontact: sip:103@172.27.1.156:5060
Peername: 103
Event: Newstate
TimeStamp: 2008-10-15 17:31:04
Privilege: call,all
Channel: SIP/103-08339fd0
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 0000000000
CallerIDName:
Uniqueid: 1224113461.572
Event: Bridge
TimeStamp: 2008-10-15 17:31:04
Privilege: call,all
Bridgestate: Link
Bridgetype: core
Channel1: DAHDI/1-1
Channel2: SIP/103-08339fd0
Uniqueid1: 1224113458.571
Uniqueid2: 1224113461.572
CallerID1: 0000000000
CallerID2: 0000000000
Event: Unlink
TimeStamp: 2008-10-15 17:31:07
Privilege: call,all
Channel1: DAHDI/1-1
Channel2: SIP/103-08339fd0
Uniqueid1: 1224113458.571
Uniqueid2: 1224113461.572
CallerID1: 0000000000
CallerID2: 0000000000
Event: Hangup
TimeStamp: 2008-10-15 17:31:08
Privilege: call,all
Channel: SIP/103-08339fd0
Uniqueid: 1224113461.572
CallerIDNum: 0000000000
CallerIDName:
Cause: 16
Cause-txt: Normal Clearing
Event: Dial
TimeStamp: 2008-10-15 17:31:08
Privilege: call,all
SubEvent: End
Channel: DAHDI/1-1
UniqueID: 1224113458.571
DialStatus: ANSWER
Event: Hangup
TimeStamp: 2008-10-15 17:31:08
Privilege: call,all
Channel: DAHDI/1-1
Uniqueid: 1224113458.571
CallerIDNum: 0000000000
CallerIDName: XXXXXXXXXXXXXX
Cause: 16
Cause-txt: Normal Clearing
And as well for anyone who might want to look at it. The 1.6 manager API doc for the changes that are in the new API:
- Manager version changed to 1.1
- CHANGED EVENTS AND ACTIONS
-
The Hold/Unhold events
- Both are now “Hold” events
For hold, there’s a “Status: On” header, for unhold, status is off - Modules chan_sip/chan_iax2
- Both are now “Hold” events
-
The Ping Action
- Now use Response: success
- New header “Ping: pong”
-
The Events action
- Now use Response: Success
- The new status is reported as “Events: On” or “Events: Off”
-
The JabberSend action
- The Response: header is now the first header in the response
- now sends “Response: Error” instead of “Failure”
-
Newstate and Newchannel events
- these have changed headers
"State" -> ChannelStateDesc Text based channel state
-> ChannelState Numeric channel state - The events does not send “” for unknown caller IDs just an empty field
- these have changed headers
-
Newchannel event
- Now includes “AccountCode”
-
Newstate event
- Now has “CalleridNum” for numeric caller id, like Newchannel
- The event does not send “” for unknown caller IDs just an empty field
-
Newexten and VarSet events
- Now are part of the new Dialplan privilege class, instead of the Call class
-
Dial event
- Event Dial has new headers, to comply with other events
- Source -> Channel Channel name (caller)
- SrcUniqueID -> UniqueID Uniqueid
(new) -> Dialstring Dialstring in app data
-
Link and Unlink events
- The “Link” and “Unlink” bridge events in channel.c are now renamed to “Bridge”
- The link state is in the bridgestate: header as “Link” or “Unlink”
- For channel.c bridges, “Bridgetype: core” is added. This opens up for
bridge events in rtp.c - The RTP channel also reports Bridge: events with bridgetypes
- rtp-native RTP native bridge
- rtp-direct RTP peer-2-peer bridge (NAT support only)
- rtp-remote Remote (re-invite) bridge. (Not reported yet)
-
The “Rename” manager event has a renamed header, to use the same
terminology for the current channel as other events- Oldname -> Channel
-
The “NewCallerID” manager event has a renamed header
- CallerID -> CallerIDnum
- The event does not send “” for unknown caller IDs just an empty field
-
Reload event
- The “Reload” event sent at manager reload now has a new header and is now implemented
in more modules than manager to alert a reload. For channels, there’s a CHANNELRELOAD
event to use.
(new) -> Module: manager | CDR | DNSmgr | RTP | ENUM
(new) -> Status: enabled | disabled - To support reload events from other modules too
- cdr module added
- The “Reload” event sent at manager reload now has a new header and is now implemented
-
Status action replies (Event: Status)
Header changes- link -> BridgedChannel
- Account -> AccountCode
- (new) -> BridgedUniqueid
-
StatusComplete Event
New header- (new) -> Items Number of channels reported
-
The ExtensionStatus manager command now has a “StatusDesc” field with text description of the state
-
The Registry and Peerstatus events in chan_sip and chan_iax now use “ChannelType” instead of “ChannelDriver”
-
The Response to Action: IAXpeers now have a Response: Success header
-
The MeetmeJoin now has caller ID name and Caller ID number fields (like MeetMeLeave)
-
Action DAHDIShowChannels
Header changes- Channel: -> DAHDIChannel
For active channels, the Channel: and Uniqueid: headers are added
You can now add a "DAHDIChannel: " argument to DAHDIshowchannels actions
to only get information about one channel.
- Channel: -> DAHDIChannel
-
Event DAHDIShowChannelsComplete
New header- (new) -> Items: Reports number of channels reported
-
Action VoicemailUsersList
Added new headers for SayEnvelope, SayCID, AttachMessage, CanReview
and CallOperator voicemail configuration settings. -
Action Originate
Now requires the new Originate privilege.
If you call out to a subshell in Originate with the Application parameter,
you now also need the System privilege.
- NEW ACTIONS
-
Action: ModuleLoad
Modules: loader.c
Purpose:
To be able to unload, reload and unload modules from AMI.
Variables:
ActionID: Action ID for this transaction. Will be returned.
Module: Asterisk module name (including .so extension)
or subsystem identifier:
cdr, enum, dnsmgr, extconfig, manager, rtp, http
LoadType: load | unload | reload
The operation to be done on module
If no module is specified for a reload loadtype, all modules are reloaded -
Action: ModuleCheck
Modules: loader.c
Purpose:
To check version of a module - if it’s loaded
Variables:
ActionID: Action ID for this transaction. Will be returned.
Module: Asterisk module name (not including extension)
Returns:
If module is loaded, returns version number of the moduleNote: This will have to change. I don't like sending Response: failure on both command not found (trying this command in earlier versions of Asterisk) and module not found. Also, check if other manager actions behave that way.
-
Action: QueueSummary
Modules: app_queue
Purpose:
To request that the manager send a QueueSummary event (see the NEW EVENTS
section for more details).
Variables:
ActionID: Action ID for this transaction. Will be returned.
Queue: Queue for which the summary is desired -
Action: QueuePenalty
Modules: app_queue
Purpose:
To change the penalty of a queue member from AMI
Variables:
Interface: <tech/name> The interface of the member whose penalty you wish to change
Penalty: The new penalty for the member. Must be nonnegative.
Queue: If specified, only set the penalty for the member for this queue;
Otherwise, set the penalty for the member in all queues to which
he belongs. -
Action: QueueRule
Modules: app_queue
Purpose:
To list queue rules defined in queuerules.conf
Variables:
Rule: The name of the rule whose contents you wish to list. If this variable
is not present, all rules in queuerules.conf will be listed.
- NEW EVENTS
-
Event: Transfer
Modules: res_features, chan_sip
Purpose:
Inform about call transfer, linking transferer with transfer target
You should be able to trace the call flow with this missing piece
of information. If it works out well, the “Transfer” event should
be followed by a “Bridge” event
The transfermethod: header informs if this is a pbx core transfer
or something done on channel driver level. For SIP, check the example:
Example:Event: Transfer Privilege: call,all TransferMethod: SIP TransferType: Blind Channel: SIP/device1-01849800 SIP-Callid: 091386f505842c87016c4d93195ec67d@127.0.0.1 TargetChannel: SIP/device2-01841200 TransferExten: 100 TransferContext: default
-
Event: ChannelUpdate
Modules: chan_sip.c, chan_iax2.c
Purpose:
Updates channel information with ID of PVT in channel driver, to
be able to link events on channel driver level.
* Integrated in SVN trunk as of May 4th, 2007
Example:
Event: ChannelUpdate
Privilege: system,all
Uniqueid: 1177271625.27
Channel: SIP/olle-01843c00
Channeltype: SIP
SIPcallid: NTQzYWFiOWM4NmE0MWRkZjExMzU2YzQ3OWQwNzg3ZmI.
SIPfullcontact: sip:olle@127.0.0.1:49054
-
Action: CoreSettings
Modules: manager.c
Purpose: To report core settings, like AMI and Asterisk version,
maxcalls and maxload settings.
* Integrated in SVN trunk as of May 4th, 2007
Example:
Response: Success
ActionID: 1681692777
AMIversion: 1.1
AsteriskVersion: SVN-oej-moremanager-r61756M
SystemName: EDVINA-node-a
CoreMaxCalls: 120
CoreMaxLoadAvg: 0.000000
CoreRunUser: edvina
CoreRunGroup: edvina -
Action: CoreStatus
Modules: manager.c
Purpose: To report current PBX core status flags, like
number of concurrent calls, startup and reload time.
* Integrated in SVN trunk as of May 4th, 2007
Example:
Response: Success
ActionID: 1649760492
CoreStartupTime: 22:35:17
CoreReloadTime: 22:35:17
CoreCurrentCalls: 20 -
Event: NewAccountCode
Modules: cdr.c
Purpose: To report a change in account code for a live channel
Example:
Event: NewAccountCode
Privilege: call,all
Channel: SIP/olle-01844600
Uniqueid: 1177530895.2
AccountCode: Stinas account 1234848484
OldAccountCode: OllesAccount 12345 -
Event: ModuleLoadReport
Modules: loader.c
Purpose: To report that module loading is complete. Some aggressive
clients connect very quickly to AMI and needs to know when
all manager events embedded in modules are loaded
Also, if this does not happen, something is seriously wrong.
This could happen to chan_sip and other modules using DNS.
Example:
Event: ModuleLoad
ModuleLoadStatus: Done
ModuleSelection: All
ModuleCount: 24 -
Event: QueueSummary
Modules: app_queue
Purpose: To report a summary of queue information. This event is generated by
issuing a QueueSummary AMI action.
Example:
Event: QueueSummary
Queue: Sales
LoggedIn: 12
Available: 5
Callers: 10
HoldTime: 47
If an actionID was specified for the QueueSummary action, it will be appended as the
last line of the QueueSummary event.
- TODO
I was able to get it working this morning. Grabbed the 1.0.0 snapshot of the asterisk-java library and then recompiled asterisk-im with that, Caller ID popups now show up with my Asterisk 1.6 system.
Oh and heres the asterisk-im plugin compiled with asterisk-java 1.0.0 snapshot. Be warned though. While it does bring things back to functionality for asterisk 1.6, the plugin config screen on openfire looks to be blank after installing it. The old config is still there for the previous version, just no way to make changes. Maybe someone else will have different results or maybe someone more familar with the plugin source can correct. This was comiled for openfire 3.6.0. --Fred
asterisk-im-server-1.4.1-SNAPSHOT.jar (551732 Bytes)