Asterisk Zimlet

[naturalblue]

Does anyone know where i can get my hands on the asterisk zimlet. i'm very intereste in trying this out.

[phoenix]

If you read the forum information you'll see it's not available yet.

[KevinH]

It's now in CVS. It's demo quality and needs more testing but feel free to try it out. This will ONLY work on mainline so from the CVS main branch. It's not going to work in the CRAY/GA branch most of you are running. There were some newer api's added in our main code line and this Zimlet uses them.

http://cvs.zimbra.com:8080/viewrep/z...imbra_asterisk

[dcheng]

Hi. This is my first post here. I hope I'm not one of those asking stupid questions. But here goes.

I've installed the development build of Zimbra 3.1 on our server. We also have an asterisk server running. After setting up the preferences I tried calling one of my officemates and it worked.

But after answering the call, what's supposed to happen next? Is the asterisk server supposed to connect my soft phone(xlite) with theirs? So far nothing seems to be happening.

here are some logs ...

** ASTERISK Logs **
== Spawn extension (default, 8123, 1) exited non-zero on 'SIP/202-0c08'
-- Executing Dial("SIP/dusty-d166", "SIP/test") in new stack
-- Called test
-- SIP/test-615d is ringing
-- SIP/test-615d answered SIP/dusty-d166
-- Attempting native bridge of SIP/dusty-d166 and SIP/test-615d



** Zimbra Tomcat Logs **
Calling from 8123 to 8124
createSipStack gov.nist.javax.sip.SipStackImpl@124614c
Binding to zimbra.mydomain.com:58023
REQUEST
INVITE sip:8124@asterisk.mydomain.com:5060 SIP/2.0
Call-ID: b4b68effe8afb534b835018041804ec5@zimbra.mydomain.c om
CSeq: 1 INVITE
From: <sip:dusty@asterisk.mydomain.com>;tag=Zimbra2570
To: <sip:8124@asterisk.mydomain.com:5060>
Via: SIP/2.0/UDP zimbra.mydomain.com:58023
Max-Forwards: 70
Contact: <sip:dusty@zimbra.mydomain.com>
Content-Type: application/sdp
Call-Info: <http://www.antd.nist.gov>
Content-Length: 145

v=0
c=IN IP4 0.0.0.0
m=audio 58023 RTP/AVP 0 8 4 18
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729A/8000

REQUEST Sent to sip:8124@asterisk.mydomain.com:5060
Response received with client transaction id gov.nist.javax.sip.stack.SIPClientTransaction@a8f7 04d8: 100
Trying number...
Response received with client transaction id gov.nist.javax.sip.stack.SIPClientTransaction@a8f7 04d8: 180
OTHER RESPONSE:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP zimbra.mydomain.com:58023;branch=z9hG4bKff3a8e46c9 1d6ec09d3db4055cf81fb1;received=172.16.1.203
From: <sip:dusty@asterisk.mydomain.com>;tag=Zimbra2570
To: <sip:8124@asterisk.mydomain.com:5060>;tag=as15202c f0
Call-ID: b4b68effe8afb534b835018041804ec5@zimbra.mydomain.c om
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTI FY
Contact: <sip:8124@zimbra.internalip>
Content-Length: 0


nulling reference

[bobby]

>I tried calling one of my officemates and it worked.
>But after answering the call, what's supposed to happen next?
>

have asterisk also call another person or your extension

[moebis]

Will this work on 3.1.1? If so, where do I put the Zimlet? I tried /opt/zimbra/apache-tomcat-5.5.15/webapps/service/zimlet/com_zimbra_asterisk/ after downloading and unzipping it. chown -R zimbra:zimbra that entire dir, now what?

[KevinH]

You need the next major release (currently called 3.5). 3.1.1 doesn't have the needed framework/api to use it.

[robbyt]

Originally Posted by KevinH

You need the next major release (currently called 3.5)...

...due later this week right??

[moebis]

Let me say it's next to impossible to get the CVS to work. I've been at it for 4 hours now. I have to use 2 documents, the http://wiki.zimbra.com/index.php?tit...e_5_and_x86_64 and http://cvs.zimbra.com:8080/viewrep/z...o.txt?r=ZIMBRA and they both have a ton of mistakes.

Why is it so difficult to compile this from source?

[KevinH]

Originally Posted by robbyt

...due later this week right??

Nope. 3.1.1 came out this week. 3.5 is later this summer.