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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.18s
  • Memory usage: 2.18MB
  • Database queries: 29
  • GZIP: Disabled
  • Server load: 3.04

Asterisk-IM - Jive Software Integration

Image

Asterisk-IM

Integration component to Asterisk for Jive's Jabber/XMPP server



Installation tips & first steps

  • you need to restart Wildfire before the Asterisk-IM plugin will work
  • in manager.conf you'll only need the "call" and "system" privileges, all others like "verbose" etc are not necessary, i.e.:
[wildfire]
secret=mypass
read=system,call
allow=123.123.123.123/255.255.255.255
  • in the "phone mapping" section you need to enter "SIP/myname" in the field labelled "Phone"
  • Asterisk-IM 1.0 and 1.1 appear not to support the Dial() action from within Spark for Asterisk 1.2 - use Asterisk 1.0 instead
  • Spark doesn't show your own status changes (if you add yourself as a contact, you can see your status change); however the admin web interface will show you as "away" when you are on the phone

Go back to Asterisk call notification

Created by muppetmaster, Last modification by gsupp on Mon 20 of Nov, 2006 [17:57 UTC]

Comments

Comments Filter

Callweaver/OpenPBX support?

by mm_202 on Wednesday 20 of June, 2007 [04:57:02 UTC]
Does anyone know if there is anyway to get Asterisk-IM to work with Callweaver (formerly OpenPBX)?

works well

by gsupp on Monday 20 of November, 2006 [17:54:46 UTC]
Wildfire 3.0.0, Asterisk-IM 1.1.1 and Asterisk 1.2.9.1 work well together. Note that Wildfire must be using an external database (such as MySQL) for Asterisk-IM to work, the internal database is not currently compatible.

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