Asterisk Wiki

Interested in talking about Mash-up's? This is the place.
gmsmith
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Postby gmsmith » Wed Aug 23, 2006 1:30 pm

[quote user="moebis"]Marcmac, this is a crock. Everyone here is having the same problem with the call not "bridging" or "originating" from the extension end. Another user fixed the problem with the configuration because of a bug, which you refused to answer or fix. "It just works" you say. Well we all only started to get this Zimlet to half work when we changed the configuration first, then rezipped the file and then Deployed the zimlet because your instructions do not work on a new install. I went through 3 reinstalls to see if I was doing something wrong, alas it was a bug, and if you had taken the time to try your instructions on a fresh install (instead of assuming everyones Asterisk server is 192.168.1.254 you would have seen that).
Now onto the new issue..... it does not work, as a matter of fact it does not for everyone in this forum thusfar, except you, the developer. Why are you using a Bridging system anyways? A similar plugin for SugarCRM simply forwards the call to the extension, and guess what? It does work, as a matter of fact it works for everyone who has ever tried to integrate Asterisk with SugarCRM.
I'm sorry if this post sounds harsh, but it does accurately convey my level of frustration about the time I/we have wasted. Reflect on an old adage Marcmac, if everyone is telling you something is wrong, and you're the only one who thinks its ok, then usually everyone else is right.[/QUOTE]
Hey now, I think you are being a little harsh. Yes, there are some kinks to be worked out, but no need to come down on Marc. He wrote a zimlet, works for the folks at Zimbra and shared the code with us. There may be some tweaks needed so we just need to be flexible a little bit. I just haven't had time to tear it down myself to see what is happening. I think there might be an easier way to implement the zimlet (the same way the SugarCRM module does it), but lets all put our heads together and see what we can come up with.
Greg


moebis
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Postby moebis » Wed Aug 23, 2006 2:54 pm

[quote user="gmsmith"]Hey now, I think you are being a little harsh. Yes, there are some kinks to be worked out, but no need to come down on Marc. He wrote a zimlet, works for the folks at Zimbra and shared the code with us. There may be some tweaks needed so we just need to be flexible a little bit. I just haven't had time to tear it down myself to see what is happening. I think there might be an easier way to implement the zimlet (the same way the SugarCRM module does it), but lets all put our heads together and see what we can come up with.
Greg[/QUOTE]
Greg, I'm glad we've at least started some sort of interest here. But really this bridging system is bizarre. I've wasted sooooooo much time trying to figure this out. I even started the wiki you now see on Asterisk, but no one other Marc has contributed, and his information for the most part is wrong. What I would like to see added to the wiki, is an explanation of the system and how it supposed to work and/or interact with your Asterisk system. Is there such a thing as a "bridge extension" I created an unused SIP extension, but still doesn't work. All it does is make my cell phone ring when I type the 10 digit number in and it never passes the call to the extension I specified.
moebis
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Postby moebis » Fri Aug 25, 2006 1:10 pm

And this topic is dead, no one else is having problems?
phoenix
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Postby phoenix » Fri Aug 25, 2006 1:28 pm

[quote user="moebis"]And this topic is dead, no one else is having problems?[/QUOTE]

Either it's dead or nobody has read this thread or they're not having problems - take your pick.
If this isn't a silly question (and it's not meant to confrontational) can't you or one of the other forum members modify the zimlet to get it working? This is, after all, meant to be a community project and I'm sure Marc has enough work to do with his normal day job.
Regards

Bill

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Per ardua ad astra
alohatone
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Postby alohatone » Fri Aug 25, 2006 5:12 pm

I am experiencing the SAME issue as everyone else.
ALL calls are "originated" from my sip account. However whether I choose bridge or mobile none work. I see asterisk acting as if my sip extension is making the call my cell phone rings and I can answer and I see a bridged call between sip/XXX and ZAP/g1/NUMBER I WANT TO CALL
It seems as if the selection of the preferences is not actually working!
I am running 4.0rc1 version of zimbra.

better start looking in /opt/zimbra/apache-tomcat-5.5.15/webapps/service/zimlet/com_zimbra_asterisk

asterisk.js and asterisk.jsp gotta be busted in there somewhere....
And yes to confirm 100% the the config command in the wiki to move the config file into place DOES NOT WORK... you manually have to go and fix it
I dunno either i dont understand what zimbra and asterisk are expecting from each other or something is broke!
alohatone
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Postby alohatone » Fri Aug 25, 2006 6:16 pm

I get this in my asterisk CLI

-- Executing Dial("SIP/300-08196640", "IAX2/binfone/18086993320||tTo") in new stack

-- Called binfone/18086993320

-- Call accepted by 144.202.243.7 (format ulaw)

-- Format for call is ulaw

-- IAX2/binfone-2 is ringing

-- IAX2/binfone-2 is making progress passing it to SIP/300-08196640

-- IAX2/binfone-2 stopped sounds

-- IAX2/binfone-2 answered SIP/300-08196640

-- Hungup 'IAX2/binfone-2'
Its as if zimbra is ACTING just like my phone instead of calling me.

I have tried dialing with a + infront of the number.

Looking in the asterisk.js (NOTE I AM NOT A JAVA PROGRAMMER)

I found this

// If it's a conf bridge, connect to the non-bridge number first

/*

if (mynum == conf) {

var tmp = to;

to = conf;

from = tmp;

}

*/

And its commented out! this seems to be the only condition checking for if mynum==conf So if i uncomment this out and I MANUUALY DIAL INTO MY conference bridge.... and then initiate the call to conference. It TRIES to dial using the conference like this
-- Created MeetMe conference 1023 for conference '33333'

-- Playing 'conf-onlyperson' (language 'en')

Aug 25 12:50:58 WARNING[12272]: chan_sip.c:6795 get_refer_info: Referred-by: Huh? Not a SIP header () Ignoring?

-- Hungup 'Zap/pseudo-1890958432'
So obviously this is NOT correct... but it seems to indicate that the prefs(right click options) may not be actually being used by the zimlet.
just trying to produce more information

I can help anyway testing possible!

[quote user="alohatone"]I am experiencing the SAME issue as everyone else.
ALL calls are "originated" from my sip account. However whether I choose bridge or mobile none work. I see asterisk acting as if my sip extension is making the call my cell phone rings and I can answer and I see a bridged call between sip/XXX and ZAP/g1/NUMBER I WANT TO CALL
It seems as if the selection of the preferences is not actually working!
I am running 4.0rc1 version of zimbra.

better start looking in /opt/zimbra/apache-tomcat-5.5.15/webapps/service/zimlet/com_zimbra_asterisk

asterisk.js and asterisk.jsp gotta be busted in there somewhere....
And yes to confirm 100% the the config command in the wiki to move the config file into place DOES NOT WORK... you manually have to go and fix it
I dunno either i dont understand what zimbra and asterisk are expecting from each other or something is broke![/QUOTE]
marcmac
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Postby marcmac » Wed Aug 30, 2006 8:20 pm

Ok, I've got a zip file attached. Just tested it. It works on my system.
Which is the only thing I've been claiming all this time.
Some questions have been asked:

Q - Why use the bridging?

A - Why not? That's how I figured out how to do this - I make no claims that it's the easiest or best way.
Q - Why not use method X,Y,Z?

A - No reason at all. Go for it.
Now - READ THIS PART:
This is not the complete zimlet, the file was to large, so I removed the jar files JainSipApi1.1.jar and nist-sip-1.2.jar. Grab these from your existing copy and recreate the zip file.
What I changed:
Modified the jsp to handle the "180 Ringing" response, which I think may be causing some of the problems seen here. That's the only logic change I made.
I added the ability to turn debugging on and off in the config_template.xml. Legal values are "true" and "false". Other values will default to "false". No value will default to a stack trace :) I've also added a limit to the number of times it will try to bind, so it won't spin if the configured IP is wrong for the local host.
How to do this:
Recreate the zip (with the two jar files.)

zmzimletctl deploy com_zimbra_asterisk.zip

zmzimletctl getConfigTemplate com_zimbra_asterisk.zip > conf.xml

edit conf.xml to set IP addresses for the local server, the sip server, and turn debug on or off.

zmzimletctl configure conf.xml

tomcat stop

tomcat start
Log in, set your accounts, and place a call.
Alternatively, just hit this URL (munged so it doesn't show up as a link):

http:/ /localhost/service/zimlet/com_zimbra_asterisk/asterisk.jsp?to=111&from=15555551212&uname=222&pass=333&debug=true
Assuming localhost is where your server is.
If debugging is turned on, output will go to /opt/zimbra/tomcat/logs/catalina.out,

and will dump all the SIP traffic.
Then, when you're frustrated, you'll at least have some data to post. And, I'll do my best to keep up with this, and work through problems.
Lastly - Moebis - attacking someone who provides free software for your enjoyment is not the correct usage of the term "open source".

com_zimbra_asterisk.zip

sakilaine
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Postby sakilaine » Thu Aug 31, 2006 6:41 am

Hello,
is necessary it to insert something of specific in the dialplan??
The URL following in file "com_zimbra_asterisk.xml" correspond to what??



http://192.168.1.254">

http://www.zimbra.comhttp://http://www.zimbra.com>


And the http://www.zimbra.com ???
The URL full it's an application in to ZIMBRA ??
Thank's

Sakilaine

http://www.asterisk-france.net
henrik_b
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Postby henrik_b » Thu Aug 31, 2006 7:02 am

[quote user="marcmac"]

Recreate the zip (with the two jar files.)

zmzimletctl deploy com_zimbra_asterisk.zip

zmzimletctl getConfigTemplate com_zimbra_asterisk.zip > conf.xml

edit conf.xml to set IP addresses for the local server, the sip server, and turn debug on or off.

zmzimletctl configure conf.xml

[/QUOTE]
still doesn't work. It still uses the 192.168.. IP instead of the IP I edited in conf.xml. Ok, I know how to handle this. Simply adjust the IP in the .zip -file before deploying it.
[quote user="marcmac"]

tomcat stop

tomcat start

[/QUOTE]

Doesn't work either, but its another problem with my Debian-install.
[quote user="marcmac"]

Log in, set your accounts, and place a call.

[/QUOTE]
OK. doesn't help either. Asterisk calls the other phone, but my desk-phone doesn't ring. Looking on the line with ethereal it seems like Zimbra is trying to transfer the call using a REFER-request but asterisk doesn't recognize that properly.

How are the extensions defined in your asterisk? Perhaps we have to search the problem there? Could you post the relevant part of your extensions.conf? Any special settings in sip.conf?
thanks Henrik
1739miguel
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Postby 1739miguel » Thu Aug 31, 2006 7:15 am

It's full workin' here! :)
Thank you very much marcmac!

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