powered by Jive Software

Only called party appears to be busy while calling party remains available

I have installed:

  • Jive Messenger v2.2.2

  • Asterisk IM Plugin v1-beta3

  • Asterisk v1.2-beta (already running for a while)

I have configured the Asterisk IM Plugin to connect to my Asterisk server successfully. I have configured my phone details as follows:

  • uname1 SIP/3002 3002 Uname1 <3002>
  • uname2 SIP/3001 3001 Uname2 <3001>

When I dial from 3001 to 3002 only 3002 appears in a call/busy when 3002 answers. If I call from 3002 to 3001 the opposite is true, as 3001 (the called party) appears in a call/busy.

Any ideas on what the problem is and how I may resolve?

I’‘m not sure if yours is one of them, but there are a couple of known presence bugs in Asterisk-IM. Andrew (the lead developer) announced that a new beta that will fix those bugs will come out soon, probably today. If you’'re lucky, your problem will be solved by simply updating your plugin.

This problem presists in beta4 as well.

I’'ve ran some tests of my own. This is the setup I used:

Jive Messenger from SVN (about two hours prior to writing)

Asterisk-IM from SVN (same)

Asterisk 1.0.9

X-Lite Softphones and Trillian with the Jabber plugin on both PCs

Two users/phone mappings (username device extension caller-id):

frank_erlang SIP/frank 3333 Frank

guus SIP/guus 2222 Guus

We’‘ve been trying these scenario’'s:

Frank calls Guus, directly with softphone.

Guus calls Frank, directly with softphone.

Frank calls Guus, via Trillian.

Guus calls Frank, via Trillian.

In all cases both Guus and Frank got a ‘‘On the Phone’’ presence in Trillian, as soon as the connection was set up.

Caller-ID failed though. In all cases, Trillian gives the receiver a popup saying: “Incoming call from …” where the dots are the extention of the receiver himself.

I have:

  • Jive v2.3beta

  • Asterisk-IM v1.0beta

  • Asterisk v1.2beta

I have installed two seperate instances, one a fairly complex configuration I have been using sometime and a fresh server and configuration. On both I had the same result, when a call was made between two endpoints on the same Asterisk instance (both configured in Asterisk-IM mapped to the appropriate users) only the called party would show as busy. If the call went the other way once again only the called party would show buy. In both cases the calling party continues as available/green in the Jive admin interface as well as in the Jabber clients.

An issues with v1.2 of Asterisk?

MuppetMaster,

Could you do the following for me:

  1. Login into asterisk manager interface via telnet.

telnet

  1. Perform the scenerio you described above.

  2. Save all the output to a file and attach it to this thread.

I can use this to see if the same events are being sent with version 1.2.

thanks,

Andrew

Could you also attach your sip.conf (You can remove passwords, etc.)

REMOVED

Message was edited by:

MuppetMaster

Manager output:

  • 50008 -> 50001

Event: Newchannel

Privilege: call,all

Channel: SIP/50008-3b9a

State: Ring

CallerID: 50008

CallerIDName: Jason Goecke

Uniqueid: 1129911103.62

Event: Newexten

Privilege: call,all

Channel: SIP/50008-3b9a

Context: jason_dialtone

Extension: 50001

Priority: 1

Application: Ringing

AppData:

Uniqueid: 1129911103.62

Event: Newexten

Privilege: call,all

Channel: SIP/50008-3b9a

Context: jason_dialtone

Extension: 50001

Priority: 2

Application: Wait

AppData: 2

Uniqueid: 1129911103.62

Event: Newexten

Privilege: call,all

Channel: SIP/50008-3b9a

Context: jason_dialtone

Extension: 50001

Priority: 3

Application: Answer

AppData:

Uniqueid: 1129911103.62

Event: Newstate

Privilege: call,all

Channel: SIP/50008-3b9a

State: Up

CallerID: 50008

CallerIDName: Jason Goecke

Uniqueid: 1129911103.62

Event: Newexten

Privilege: call,all

Channel: SIP/50008-3b9a

Context: jason_dialtone

Extension: 50001

Priority: 4

Application: Dial

AppData: SIP/50001&IAX2/50001|30|tT

Uniqueid: 1129911103.62

Event: Newchannel

Privilege: call,all

Channel: SIP/50001-32a2

State: Down

CallerID:

Uniqueid: 1129911105.63

Event: Link

Privilege: call,all

Channel1: SIP/50008-3b9a

Channel2: SIP/50001-32a2

Uniqueid1: 1129911103.62

Uniqueid2: 1129911105.63

CallerID1: 50008

CallerID2: 50001

Event: Unlink

Privilege: call,all

Channel1: SIP/50008-3b9a

Channel2: SIP/50001-32a2

Uniqueid1: 1129911103.62

Uniqueid2: 1129911105.63

CallerID1: 50008

CallerID2: 50001

Event: Hangup

Privilege: call,all

Channel: SIP/50001-32a2

Uniqueid: 1129911105.63

Cause: 16

Cause-txt: Normal Clearing

Event: Hangup

Privilege: call,all

Channel: SIP/50008-3b9a

Uniqueid: 1129911103.62

Cause: 16

Cause-txt: Normal Clearing

I have to apologize for wasting people’'s time here. I re-ran the test with a different endpoint configuration.

I was originally running Fire on OSX and Trillian inside of a Windows XP/Virtual PC session on the same machine. Turns out it appears the packets collide to the two endpoints on the same machine (problems with Virtual PC networking) and therefore did not reach each endpoint.

I seperated the test to Fire on OSX and Trillian on a seperate Windows XP machine and all works fine.

So, recommendation, do not try two endpoints on an OSX platform and Virtual PC on the same box! (Not that anyone else would have tried something like this…)

I am having a similiar issue; when calling an inter office extension the called party appears busy but the calling party remains availabile. However, if I call out on an IAX line the calling party appears busy, as they should.

What type of data can I send you to help solve this?

  • Asterisk CVS-v1-0-10/30/05-21:22:19

  • jive_messenger_2_3_0_beta_1.rpm

  • asterisk-im.jar 1.0 Beta 4

I upgraded Asterisk to CVS-Nv1-2-0-beta1-10/31/05-00:19:53, same issue. However, up in the “My Phone Calls” area it shows me making an “Outgoing call to…”

Also, the incoming caller ID isn’'t correct. It always shows the extension of the called party.